SBC ( Session Borde Controllers ) are basically gateways that provide interconnectivity between the hosted IP-PBX of the enterprise to the outside world endpoints such as telco service provider, PSTN/ TDM , SIP trunking providers or even third party OTT provider apps like skype for business etc.
If you have a hosted IPPBX or PBX in your data-centre or on premise and you need controlled but heavy outflowing traffic, it is a good idea to integrate a resilient and efficient SBC to provide seamless interconnectivity.
Hosted PBX
For an enterprises such as an Trading floor or warehouse with multiple phone types , softphones , hardphones , turrets etc distributed across various geographies and zones a device agnostic architectural setup is prime . Listing the essentials for setting up such a system. Note supplementary services are data-services , logging , licensing etc are important but kept out of scope to keep focus on functional aspects .
An enterprise application usually is structured in tiers or layers
Client tier – the networks clients communication to the central java programs . Runs on client machines
web tier – state full communication between client and business tier . Runs in server machine.
business tier- handles the logic of the application. The business tier uses the Enterprise Java Bean (EJB) container, which manages the execution of the beans
data tier – encompasses DB drivers . Runs on separate machines for database storage
Event services for Line status notifications
providers lines status notification across enterprise for inter zone and softphone to hardphone .
Routing services
routing calls within enterprise and hardphone sites read more about resource zones later in the article
Call Control Manager (CCM)
consolidated set of all service and component that make up the VOIP platform besides media handlers . It includes SIP adapters , bridge managers , call processing frameworks , API frameworks , healthchecks etc .
Call processing framework ( CPF)
signalling and call routing logic , mostly in SIP and trunks . Manages identities such as Call Line information , Called Party Information , line status etc in shared memory.
Multiple shared Lines and their statuses
Incases where there is a need to process multiple calls from a single User agent device such as a softphone or hardphone ( common scenario for a turret phone) , the design involves assigning it multiple sip uris and each sip uri will establish a line.
When caller calls callee , the line is said to be BUSY , otherwise said to be IDLE. Transition of a shared sip line from IDLE to BUSY is transmitted to others via SIP PUBLISH as other UAs holding the same sip
Similarly any other event like transfer is propagated to other via SIP UPDATE
Clustering Call control managers (CCM)
A Call Communication manager (CCM) from various zones should be able to cowork on call and session management and advanced features such as routing from home guest zone to home zone , call transfer , refer , barge etc. Designing a clustered setup will also provide elasticity , fail-over and high availability. Can use clustered , HA compliant framework such as Oracle Communication Application Server , suited for enterprise level deployments.
Call Replication and distributed memory management
A node will store two types of data: active sessions and passive sessions. The active sessions are used by the node and stored in cache. The passive sessions are the replicas from the other nodes’ active sessions. The passives sessions are stored on a persistent storage.
Controlling Line Calls using AOR and Resource Zones
When dealing with many SIP endpoints , now referred to as resource, it is best to assign the resources to their respective zones. Thus a resource’s status updates will be only updated by its active resource zone while can be read by any resource zone.
Incoming request Zone vs Active Resource Zone
For an Incoming request such a INVITE , check whether the zone sending the request is its active resource zone or not .If the Active Resource Zone is the same zone on which the INVITE came in, then the call is handled by that zone. If the Active Resource Zone is a different zone, then the call needs to be forwarded to the Active Resource Zone.
Bridges for Local Media connections
Although call signalling is handled by a resources active resource zone only, we can still create media bridges in local zone of the resource .
Local MM bridges are used to auto answer an incoming sip line call and create trunk , especially from hardphones which do not support provisional responses.
Interzone proxy Handler
proxies call control messages between active and non active resource zones. Primarily mapping the sip messages with all custom headers inbetween the communication device interfaces.
Dial Trunk using multiple dedicated sip lines and connect via Media Bridge
To save up on call routing /connection time and to support te ability to add as many users on call at runtime , a dedicated media bridge is established for every call.
A sip line activated is auto-answered by MM , creates a trunk and waits for other endpoint to join the bridge. The flow is as follows :
As INVITE arrives for an IDLE sip line , it is connected to a trunk and auto answered by a local MM bridge .
Since the call is already answered , when caller dials number for callee , collect the DTMF digits over RTP using RFC 2833 DTMF events.
Run inter-digit timer for digit collection and detect end of dialing on timeout.
The dialed trunk connection is made and call is added to media bridge
When provisional responses are received on the trunk connection, generate in-band call progress tones (ringing, proceeding etc) via the MM
When the line answers, the progress tones have to be stopped and the called party gets bridged to the calling party via the media bridge.
Call Diversion involves forwarding calls from zone to another zone. joinjed parties get call UPDATE status and forward response .
Call barge is the processing of joining an ongoing call . The barge event is usually propagated to joined parities via SIP INFO. Private lines do not allow barge in and are exclusively reserved for only few users.
Interconnectivity provided by an SBC ( Session Border Controller)
Hold-Resume and Music on Hold in multi-line evironment
While a regular p2p call involves simple reinvite based hold and resume with varrying SDP, the scenario is slightly more detailed for hold resume on bridged trunk connection , as explained below.
As the calls made are on bridge , a hold signal involves a RE-INIVITE with held-SDP to media manager (MM). If hold status on trunk is 200 OK the hold status will be sent to other call interfaces connected on the trunk. Else if hold is denied ,403 is sent back to hold-initiates.
Music on hold is an one way RTP mostly from media server.
For a bridged scenarios , separate Music on hold bridges are kept on Media Managers. When an UA has to hold , it is removed from original bridge and place on music on hold bridge . To be unhold/ resume it is placed back into the orignal bridge from music on hold bridge .
Conference
user initiates conference, the conference feature can execute on the zone where the user was logged on, irrespective of zones where the other conference attendees join from . The Call processing framework of originators zone completes the SDP exchange to establish two-way speech path among all the parties.
Incases there are multiple connections from a zone , a local MM conference bridge can be created for them which would connect back to originators MM conf bridge . this two part conf bridge will be transparent to the sip line sand users .
For provisioning inputs and settings setup a Diagnostics , Administration and Configuration platform which can process APIs for data services , licences , alarms or do remote device control such as using SNMP
Session Border Controllers (SBC)
At network level SBC operations include
bridging multiple interfaces in different networks even between the IPv4 and IPv6 networks
auto NAT discovery and STUN
protocol conversion such as TLS to UDP etc
Flood detection and IP filtering
For SIP specific functionalities , SBC does
SIP validation involving checks on syntax and message contents also consistency checks are performed.
stateful and call aware. tracing, monitoring and checking for validitya and health of all the SIP messages
Topology hiding
Traffic filtering
Codec filtering , reordering , media pinning, transcoding, or call recording
Data replication brings High Availability (HA) with hot backups or even Active-Active solutions.
Traffic sharing and routing roles of SBC can include
IP-based and Digest-based authentication
limiting traffic by number of concurrent calls or calling rate.
Dialplan and/or Custom routing
Dispatching/Load-balancing to a backend cluster of servers
SBC’s can be physical hardware boxes or software based applications, as the name suggests their purpose is to control the session at border between the enterprise and external service provider.
SIP to PSTN – SIP is an IP protocol whereas PSTN is a TDM one , achieving interoperability is also the KRA of an SBC
SIP trunking – SBC provide a secure sip connectivity to connect calls to sip trunks which provide bulk calls functionality at a flat pricing.
support for various fixed or mobile endpoints – SBC ensure they are RFC compliant and can extend SIP to any kind of telecom endpoint like PSTN , GSM, fax , Skype , sipphone , IP phones etc.
NAT / Network address translator – To meet the packet routing challenges across a firewall or even during private -public mapping. A combo of DHCP servers and NAT provider comes very handy to reroute or perform hole punching such that signalling and media packets are not dropped and meet the required endpoint. More about NAT here – NAT traversal using STUN and TURN.
Load balancing – Reverse proxies and Load balancers is a much adopted industry practise to mask the inner IPs of the VoIP platform and also route traffic appropriately between control and media server .
Security , QoS and Regulatory compliance – since SBCs are required to typically support a large array of clients they adhere to regulatory and industry accepted standards ,which also involves security features like AAA, TLS/SSL and other means for quality of assurance like logging and fault detection, preventing DDoS etc . In many cases SBC can also encrypt / decrypt RTP streams for probing , tapping or lawful inspection .
Inaddition to above it is good to have if an SBC provides extra features like forking , emergency number dialing ( 911 ) or active directory integration . Real Time Analysis and monitoring of call and metrics are also expected from a SBC since they reside on edge of the network and are more vulnerable to threats . For example Dialogic Mediant SBC’s and gateways , Audio Codes SBCs
With the shift from on premise PBXs to cloud based VM or microservice architecture , SBC vendors adopt a lager umbrella of services also including automation scripts for checks , reporting tools / consoles , developer friendly APIs to manage sessions via SBC and even WebRTC gateways to connect browser endpoints .
Usage Scenarios
Any VOIP dependant system which deals with bulksome voice / video traffic from external endpoints is a usages scenarios. Listing few
Contact Call centres
Remote work / offsite monitoring
CRM solution for sales/marketing
Connecting webrtc click to dial from webpage to enterprise representatives
connecting enterprise UCC clients to PSTN endpoints
In a VOIP system, where SIP is a signaling protocol , a SIP proxy never participates in the media flow, thus it is media agnostic.
SDP packets describing a session with codecs , open ports , media formats etc are embedded in a SIP request such as invite .
Post a SDP Offer/Answer flow , RTP and RTCP esnsure that mediastream flow between the endpoints .
RTP is the provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.
RTCP is the control protocl which provides monitoring of the data delivery, qos in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.
RTP
protocol framework
supports use of RTP-level translators and mixers.
independent of the underlying transport and network layers.
does not address resource reservation
does not guarantee quality-of-service for real-time services.
services like payload type identification, sequence numbering, timestamping and delivery monitoring.
The sequence numbers included in RTP allow the receiver to reconstruct the sender’s packet sequence,
Usage :
Multimedia Multi particpant conferences
Storage of continuous data
Interactive distributed simulation
active badge, control and measurement applications
Simple Multicast Audio Conference
Assume obtaining a multicast group address and pair of ports. One port is used for audio data, and the other is used for control (RTCP) packets.
The audio conferencing application used by each conference participant sends audio data in small chunks of ms duration.
Each chunk of audio data is preceded by an RTP header; RTP header and data are in turn contained in a UDP packet.
The RTP header indicates what type of audio encoding (such as PCM, ADPCM or LPC) is contained in each packet so that senders can change the encoding during a conference, for example, to accommodate a new participant that is connected through a low-bandwidth link or react to indications of network congestion.
Every packet networks, occasionally loses and reorders packets and delays them by variable amounts of time. Thus RTP header contains timing information and a sequence number that allow the receivers to reconstruct the timing produced by the source.
The sequence number can also be used by the receiver to estimate how many packets are being lost.
For QoS, each instance of the audio application in the conference periodically multicasts a reception report plus the name of its user on the RTCP(control) port. The reception report indicates how well the current speaker is being received and may be used to control adaptive encodings. In addition to the user name, other identifying information may also be included subject to control bandwidth limits.
A site sends the RTCP BYE packet when it leaves the conference.
Audio and Video Conference
Audio and video media y are transmitted as separate RTP sessions . separate RTP and RTCP packets are transmitted for each medium using two different UDP port pairs and/or multicast addresses.
There is no direct coupling at the RTP level between the audio and video sessions, except that a user participating in both sessions should use the same distinguished (canonical) name in the RTCP packets for both so that the sessions can be associated.
synchronized playback of a source’s audio and video is achieved using timing information carried in the RTCP packets
Mixers , Translators and Monitors
Mixer
An intermediate system that receives RTP packets from one or more sources, possibly changes the data format, combines the packets in some manner and then forwards a new RTP packet.
example of Mixer for hi-speed to low-speed packet stream conversion
In conference cases where few participants are connected through a low-speed link where other have hi-speed link, instead of forcing lower-bandwidth, reduced-quality audio encoding for all, an RTP-level relay called a mixer may be placed near the low-bandwidth area.
This mixer resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower-bandwidth packet stream across the low-speed links.
All data packets originating from a mixer will be identified as having the mixer as their synchronization source.
The RTP header includes a means for mixers to identify the sources that contributed to a mixed packet so that correct talker indication can be provided at the receivers.
Translator
An intermediate system that forwards RTP packets with their synchronization source identifier intact.
Examples of translators include devices that convert encodings without mixing, replicators from multicast to unicast, and application-level filters in firewalls.
Tranasltor for Firewall Limiting IP packet pass
Some of the intended participants in the audio conference may be connected with high bandwidth links but might not be directly reachable via IP multicast, for reasons such as being behind an application-level firewall that will not let any IP packets pass. For these sites, mixing may not be necessary, in which case another type of RTP-level relay called a translator may be used.
Two translators are installed, one on either side of the firewall, with the outside one funneling all multicast packets received through asecure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the site’s internal network.
Other cases :
video mixers can scales the images of individual people in separate video streams and composites them into one video stream to simulate a group scene.
Translator usage when connection of a group of hosts speaking only IP/UDP to a group of hosts that understand only ST-II, packet-by-packet encoding translation of video streams from individual sources without resynchronization or mixing.
Monitor
An application that receives RTCP packets sent by participants in an RTP session, in particular the reception reports, and estimates the current quality of service for distribution monitoring, fault diagnosis and long-term statistics.
Layered Encodings
In conflicting bandwidth requirements of heterogeneous receivers, Multimedia applications should be able to adjust the transmission rate to match the capacity of the receiver or to adapt to network congestion.
Rate-adaptation should be done by a layered encoding with a layered transmission system.
In the context of RTP over IP multicast, the source can stripe the progressive layers of a hierarchically represented signal across multiple RTP sessions each carried on its own multicast group. Receivers can then adapt to network heterogeneity and control their reception bandwidth by joining only the appropriate subset of the multicast groups.
RTP Session
In an RTP session, each particpant maintains a full, separate space of SSRC identifiers. The set of participants included in one RTP session consists of those that can receive an SSRC identifier transmitted by any one of the participants either in RTP as the SSRC or a CSRC or in RTCP.
32-bit numeric SSRC identifier for source of a stream of RTP packets.
All packets from a synchronization source form part of the same timing and sequence number space, so a receiver groups packets by synchronization source for playback.
the binding of the SSRC identifiers is provided through RTCP.
If a participant generates multiple streams in one RTP session, for example from separate video cameras, each MUST be identified as a different SSRC.
Contributing source (CSRC)
A source of a stream of RTP packets that has contributed to the combined stream produced by an RTP mixer.
The mixer inserts a list of the SSRC identifiers of the sources , called CSRC list, that contributed to the generation of a particular packet into the RTP header of that packet.
An example application is audio conferencing where a mixer indicates all the talkers whose speech was combined to produce the outgoing packet, allowing the receiver to indicate the current talker, even though all the audio packets contain the same SSRC identifier (that of the mixer).
RTCP
periodic trnsmission of control packet
underlying protocol must provide multiplexing of the data and control packets
-provide feedback on the quality of the data distribution , congestion control , fault dialoginis , control of adaptive encodings
-carries a persistent transport-level identifier for an RTP source called the canonical name or CNAME , which is used to keep track of each participant
-observer number of particpants to rate of senidng packets for scaling up
-convey minimal session control information
Exmample of RTCP sender and receiver reports on transmission and reception statistics
Real-time Transport Control Protocol (Receiver Report)
[Stream setup by SDP (frame 4)]
[Setup frame: 4]
[Setup Method: SDP]
10.. .... = Version: RFC 1889 Version (2)
..0. .... = Padding: False
...0 0001 = Reception report count: 1
Packet type: Receiver Report (201)
Length: 7 (32 bytes)
Sender SSRC: 0x796dd0d6 (2037240022)
Source 1
Identifier: 0x00000000 (0)
SSRC contents
Fraction lost: 0 / 256
Cumulative number of packets lost: 1
Extended highest sequence number received: 6534
Sequence number cycles count: 0
Highest sequence number received: 6534
Interarrival jitter: 0
Last SR timestamp: 0 (0x00000000)
Delay since last SR timestamp: 0 (0 milliseconds)
In RTP, multiplexing is provided by the destination transport address (network address and port number) which is different for each RTP session ( seprate for audio and video ). This helps in cases where there is chaneg in encodings , change of clockrates , detection of packet loss suffered and RTCP reporting .
Moreover RTP mixer would not be able to combine interleaved streams of incompatible media into one stream.
Interleaving packets with different RTP media types but using the same SSRC would introduce several problems.
But multiplexing multiple related sources of the same medium in one RTP session using different SSRC values is the norm for multicast sessions.
To resolve hostname into ips it can do either of below
use libresolv and a combination of the locally configured DNS server /etc/hosts and the local Network Information Service (NIS/YP a.s.o)
or cache the query results and first look into internal cache
DNS failover – if destination resolves to multiple addresses tm can try all of them until it finds one to which it can successfully send the packet or it exhausts all of them , with internal DNS cache. Also used when the destination host doesn’t send any reply to a forwarded invite within the SIP timeout interval (tm fr_timer parameter).
DNS load balancing – SRV based load balancing with weight value in the DNS SRV record.
Drawbacks
Only the locally configured DNS server (usually in /etc/resolv.conf) is used for the requests (/etc/hosts and the local Network Information Service are ignored). Workaround: disable the DNS cache (use_dns_cache=off or compile without -DUSE_DNS_CACHE).
The DNS cache uses extra memory Workaround: disable the DNS cache.
The DNS failover introduces a very small performance penalty Workaround: disable the DNS failover (use_dns_failover=off).
The DNS failover increases the memory usage (the internal structures used to represent the transaction are bigger when the DNS failover support is compiled). Workaround: compile without DNS failover support (DUSE_DNS_FAILOVER).Turning it off from the config file is not enough in this case (the extra memory will still be used).
NAT ( Network Address Translation)
Network address translation replaces the IP address within packets with a different IP address which internet endpoints can relate with Enables multiple hosts in a private subnet with their pwn private address ( 10.x.x.x or 192.x.x.x etc ) to share single public IP address interface, to access the Internet.
NAT ( Network Address Translation)
NAT is bidirectional- If the private ip:port got translated to public ip:port on the inside interface while entering outside internet, on arriving from outside interface it will get translated from public ip:port to private ip:port
For a SBC ( Session border controller ) or where the kamailio server is directly customer facing , where you dont have a private line or VPN to clients, then it is often encountered with NATed endpoints. Read more about NAT traversal using STUN and TURN here
Why is Nat important in SIP?
These characteristics of SIP design and operation flows demonstrate why NAT solutions are so important ,
RFC 3261 for SIP presumed end-to-end reachability and does not specify much around ANT issues .
No NLRI (Network Layer Reachability Information) translation layer exists, such as DNS or ARP
SIP is designed to used RTP which uses dynamically allocated ports to stream media. It is comparable to FTP which creates ephemeral connections on unpredictable dynamic ports to send multiplexed data and “metadata”, instead of protocol like HTTP where all data is sent on same connection.
UDP (default transport for SIP) is connection less and session tracking requires these be mapped onto a statelful flow, rigorous keepalives and other such techniques like using TCP instead have their own tradeoffs
since sip packets put network and transport information right on sip header they are limited by the rateability and awareness of their network interface thereby prevent other endpoint from reaching its ip or port
Types of NAT solutions
Client-side NAT traversal – clients are responsible for identifying their WAN NLRI and adding ip and port to navigate them in outside world
Server-side NAT traversal – SIP server should discover the client’s WAN addressing while clients continue to work transparently behind NAT. Requires that DIP server look at the source and destination ip and port of actual packets instead of relying on the encapsulated sip headers and SDP body.
ALG (Application Layer Gateways) – mostly applied at router itself. wodk by susbtitung public IP/port information inplace of provate and vice versa for return packets . Limitataions – they dont provide a fullproof fix example they may fix Via but not the Contact address or SDP body or RTP ports
NAT behaviours
Cone NAT
Local client performs an outbound connection to a remote UA and a dynamic rule is created for the destination IP tuple, allowing the remote machine to connect back. Further subdivied into: – Full Cone NAT – Restricted Cone NAT – Port-Restricted Cone NAT
Symmetric NAT
Local client allows inbound connections from a specific source IP address and port, also NAT assigns a new random source port for each destination IP tuple
NAT behaviours
Cone NAT
Local client performs an outbound connection to a remote UA and a dynamic rule is created for the destination IP tuple, allowing the remote machine to connect back. Further subdivied into: – Full Cone NAT – Restricted Cone NAT – all requests from the same internal IP address and port are mapped to the same external IP address and port. – Port-Restricted Cone NAT
Symmetric NAT
Local client allows inbound connections from a specific source IP address and port, also NAT assigns a new random source port for each destination IP tuple
RTP NAT
NAT not only applies to sip signalling packets but also to RTP. Even SIP packets are abel to transverse accross private -public network interfaces to the right place across a NAT’d connection, that doesn’t solve two-way media.
RTP performs RTP latching where client listens for at least one RTP frame arriving at the destination port it advertised, and harvests the source IP and port from that packet and uses that for the return RTP path. RTP latching works out of the box for puclin RTP endpoints but not for ones behind NAT.
It is thus recommended to use an intermediate RTP relay such as RTPengine on kamailio. It is controlled via a UDP control socket by kamailio as an external process. More on installation and descrition of RTP engine on kamailio is covered here. When RTPengine control module receives RTP offer /answer from akmailio , it opens a pair of RTP/RTCP ports to receive traffic and substitues in SDP. Doing so for both ends makes RTP engine come in media stream packets of both directions
Fixing NAT
when the client is behind NAT, following needs to be taken careof to provide smooth operation
Ensuring Tranactional replies are sent to correct source address ( maybe using ;rport param and forcerport() method ) instead of just relying on via header transport protocol and port.
example:
Any far-end NAT traversal solution ( TURN server) if employed should stay i path of entire Dialog not just for initial INVITE transaction which many times results in ACK being dropped. This can be achived by adding Record-Route header of rr module to the initial INVITE request itself
set the advertised address of the public-facing inetrface to the Public NAT IP using “listen” parameter
Ensure contact URI is NAT processed by using NATHelper modules which rewrites the domain portion of the Contact URI to contain the source IP and port of the request or reply. add_contact_alias([ip_addr, port, proto]) in NAThelper module which adds “;alias=ip~port~transport” parameter to the contact URI containing either received ip, port, and transport protocol or those given as parameters , so
Contact:
is turned into:
Contact:
implement RTP proxy which performs NAT for streams such as rtpengine module
NAT Traversal Module
Provides far-end NAT traversal to kamailio’s SIP signalling . Its role is
detect user agents behind NAT
manipulate SIP headers so that user agents can continue working behind NAT transparently
keepalives to UA behind NAT to preserve their visibility in network
pros
even detect UAs behind multiple cascaded NAT boxes, complex distributed env with multiple proxies
handle env where incoming and outgoing paths are diff for SIP messages
handle cases when routing path may even change between consecutive dialogs
can work for other than registered UA’s also
cons
built for IPv4 NAT handling not adapted to support IPv6 session keepalives.
Why use keepalive when Registrations are already there for NATing ?
NAT binding works for registered users who want incoming calls. However for cases like outgoing calls or for presence subscription notifications, failings registration implies inability to receive further in-dialog messages after the NAT binding expires. This artificial binding for registrations makes system unreliable and volatile as it doesnot guarantee the delivery of in-dialog messages for outgoing calls without registration renewal. Therefore keepalive are adopted which also works for unregistered users.
Minimizes the traffic as only border proxies send keepalives which send keepalives statelessly, instead of having to relay messages generated by the registrars.
Also for situations when DNS resolves diff proxies for outgoing or incoming path traditional register based keepalives fail to associate or dissociate correct routes.
How keepalives work for NATing ?
This mechanism works by sending a SIP request to a user agent behind NAT to make that user agent send back a reply. The purpose is to have packets sent from inside the NAT to the proxy often enough to prevent the NAT box from timing out the connection.
Module sends Keeplaives to preserve their visibility only in :
Registration – for user agent that have registered to for incoming calls, triggering keepalive for a REGISTER request.
Subscription – for presence agents that have subscribed to some events for receiving back notifications with SUBSCRIBE request.
Dialogs – for user agents that have initiated an outgoing call for receiving further in-dialog messages. When all the conditions to keepalive a NAT endpoint will disappear, that endpoint will be removed from the list with the NAT endpoints that need to be kept alive.
function nat_keepalive() :
the function needs to be called on proxy directly interacting with UA behind NAT.
call only once for the requests (REGISTER, SUBSCRIBE or outgoing INVITEs) that triggers the need for network visibility.
call before the request gets either a stateless reply or it is relayed with t_relay()
for outgoing INVITE , it triggers dialog tracing for that dialog and will use the dialog callbacks to detect changes in the dialog state.
Dependencies – sl , tm and dialog module
Params
keepalive_interval – time interval between sending a keepalive message to all the endpoints that need being kept alive. A negative value or zero will disable the keepalive functionality.
keepalive_extra_headers – extra headers that should be added to the keepalive messages. Header must also include the CRLF (\r\n) line separator. Multiple headers can be specified by concatenating with \r\n separator.
keepalive_state_file – filename where information about the NAT endpoints and the conditions for which they are being kept alive is saved . It is used when Kamailio starts to restore its internal state and continue to send keepalive messages to the NAT endpoints that have not expired in the meantime. Also used at kamailio restart as it avoids losing keepalive state information about the NAT endpoints.
client_nat_test – Check if the client is behind NAT. Tests to be performed gievn by int can be : 1 – tests if client has a private IP address or one from shared address space in the Contact field of the SIP message. 2 – tests if client has contacted Kamailio from an address that is different from the one in the Via field. 4 – tests if client has a private IP address or one from shared address space in the top Via field of the SIP message.
For example calling client_nat_test(“3”) will perform test 1 and test 2 and return true if at least one succeeds, otherwise false.
fix_contact() – replace the IP and port in the Contact header with the IP and port the SIP message was received from. Usually called after a succesfull call to client_nat_test(type)
if (client_nat_test("3")) {
fix_contact();
}
nat_keepalive() – Triggers keepalive functionality for the source address of the request. When called it only sets some internal flags, which will trigger later the addition of the endpoint to the keepalive list if a positive reply is generated/received (for REGISTER and SUBSCRIBE) or when the dialog is started/replied (for INVITEs). For this reason, it can be called early or late in the script. The only condition is to call it before replying to the request or before sending it to another proxy. If the request needs to be sent to another proxy, t_relay() must be used to be able to intercept replies via TM or dialog callbacks.
If stateless forwarding is used, the keepalive functionality will not work. Also for outgoing INVITEs, record_route() should also be used to make sure the proxy that keeps the caller endpoint alive stays in the path.
keepalive_endpoints – total number of NAT endpoints that are being kept alive.
registered_endpoints – NAT endpoints kept alive for registrations
subscribed_endpoints – NAT endpoints kept alive for subscriptions.
dialog_endpoints – Indicates how many of the NAT endpoints are kept alive for taking part in an INVITE dialog.
NATHelper Module
NAT traversal and reuse of TCP connections Helps symmetric UAs who are not able to determine their public address.
NAT pinging types
UDP packet – 4 bytes (zero filled) UDP packets are sent to the contact address.
pros : low bandwitdh traffic, easy to generate by Kamailio;
cons : unidirectional traffic through NAT (inbound – from outside to inside); As many NATs do update the bind timeout only on outbound traffic, the bind may expire and closed.
SIP request – a stateless SIP request is sent to the UDP contact address.
pros : bidirectional traffic through NAT, since each PING request from Kamailio (inbound traffic) will force the SIP client to generate a SIP reply (outbound traffic) – the NAT bind will be surely kept open.
cons : higher bandwitdh traffic, more expensive (as time) to generate by Kamailio;
Dependencies – usrloc
Params
force_socket – Socket to be used when sending NAT pings for UDP communication.
natping_interval
ping_nated_only
natping_processes – How many timer processes should be created by the module for the exclusive task of sending the NAT pings.
natping_socket
received_avp – AVP) used to store the URI containing the received IP, port, and protocol by fix_nated_register
sipping_bflag
sipping_from
sipping_method
natping_disable_bflag
nortpproxy_str
keepalive_timeout
udpping_from_path
append_sdp_oldmediaip
filter_server_id
Functions
fix_nated_contact() -rewrites the “Contact” header field with request’s source address:port pair
fix_nated_sdp() adds the active direction indication to SDP and updates ource ip address information too
add_rcv_param() – add a received parameter to the “Contact” header fields or the Contact URI.
fix_nated_register() exports the request’s source address:port into an AVP to be used during save()
nat_uac_test()- check if client’s request originated behind a nat
is_rfc1918()
add_contact_alias() – Adds an “;alias=ip~port~transport” parameter to the contact URI
handle_ruri_alias() – Checks if the Request URI has an “alias” parameter and if so, removes it and sets the “$du” based on its value.
set_contact_alias()
Pseudo Variables
$rr_count – Number of Record Routes in received SIP request or reply.
$rr_top_count – If topmost Record Route in received SIP request or reply is a double Record Route, value of $rr_top_count is 2.
Ramudroid is an ingeniously build robot to clean outdoors and alleys inspired by Bharat Swachhata Abhiyaan . Read more
Prototype in Development
Frame and wheel
solar panel mounted
Algorithm Enhancements
Obstruction Detection
Stop Operation on Tray Weight reaching threshold
Pause operation and take Shelter in Rain
Equipment Cost
Power And Charge Devices
Solar Panel MicroSun MS 12v 60 WP – 2500 INR
Solar charger Controller – 600 INR
Battery 11.1 V 2200 mAh – 500 INR
Frame and Motion Assembly
Wheels 10 cm diameter – 50 INR x 4 ie 200 INR
Tray – 400 INR
Frame Assembly – 1000 INR
Arduino to control Motors Drivers – 500 INR
Motor Driver – 300 INR
LCD display – 200 INR
Electronics , Communicating modules and Sensors
Raspberry Pi Moddel B+ – to be replaced with low cost alternative – 2700 INR
GPS module – 700 INR
GSM module – 1400 INR
Camera 5 MP Board Module – 450 INR
Total Cost to Develop – 12000 INR
Working Principle
The robot is divided into 2 parts – Cleaning Unit and driving unit
Driving Unit
consists of 4 wheel to drive the setup . Wheels must be tightly fixed inot position to prevent them from tilting, spreading outwards and imbalancing the load. 2 rear controller by 12 V 1 amp DC motors with 300 RPM and 2 front free wheels. Motors are conneted to Arduino for receving command for start , stop , left or right navigation.
There are three input pins for each motor, including Input1 (IN1), Input2 (IN2), and Enable1 (EN1) for Motor1 and Input3, Input4, and Enable2 for Motor2.
IN1 IN2 MOTOR
0 0 BRAKE
1 0 FORWARD
0 1 BACKWARD
1 1 BRAKE
Cleaning Unit
Uses 3 tough bristled brushes controlled by 3 5V DC gear motors with 60 RPM. The arrangement of the brushes is such that the bottom 2 brushes use clockwise and anticlockwise motion outwards to pull in the litter and push up with the flow of the air and bristles of the brush. The third brush combs the collected into the collector tray. The tray is attached to weight control system to stop operations when critical weight is reached to prevent overloading the robot
Solar specification
Maximum Power (Pmax) – 60 Wp
Voltage at Maximum Power (Vmpp) – 18.1 V
Current at Maximum Power (Impp) – 3.32 A
Open Circuit Voltage (Voc) – 22.32 V
Short Circuit Current (Isc) – 3.63 A
Standard Test Conditions (STC): air mass AM 1.5, irradiance 1000W/m2, cell temperature 25°C
Maximum System Voltage 1000 V
Electrical Data at NOCT
Temperature – 47±2 °C
Nominal Operating Cell Temperature (NOCT): 800W/m2, AM 1.5, windspeed 1m/s,ambient temperature 20°C
Thermal Ratings
Operating Temperature Range -20~90 °C
Temperature Coefficient of Pmax -0.43 %/°C
Temperature Coefficient of Voc -0.36 %/°C
Temperature Coefficient of Isc 0.66 %/°C
Material Data
Panel Dimension (H/W/D) 705x655x35 mm
Weight 6 kg
Cell Type Polycrystalline
Cell Size 156×156 mm
Cell Number 36
Encapsulant Type – EVA ( Ethylene vinyl acetate)
Frame Type Anodized Aluminium Alloy
Physical
Dimentions – 70mm x 655mm x 35mm
cells per module – 36
cell type – poly crystalline sIlicon
fuel cell dimention – 156mm x 156mm
Encapsulation – EVA
back cover – PV sheet
Solar panel weight – 5kg with annodixed alumium frame , 3 kg without the frame with just the toughened texture glass on panel Frame and wheels – 2kg Accessories – 1 kg garbage holding capacity – 2 kg Total Weight of the Robot : maximum upto 10Kg
Scenarios
Good Sunlight scenario : This 12 Volt solar panel provide about 2.5 Amps of current on average during daytime. In such a situtaion it is directly used to drive the machine’s motors for wheels and brushes and electrical components such as PI and arduino. In no motion of rest conditions the genrated power is used to charge the attached backup battery.
Shady / evening / morning scenario : When the panel is not receiving direct or strong sunlight, the power generated is less hence not sufficient to take the load of driving the wheels for movement. Hence is the power falls below a certain prespecified threhold, the current is drawn from battery backup.
Night / No sunlight Scenario : battery is used to power the setup. panel can be dismounted to lower the load.
Contributing to Ramudroid Project or Reuse
It is deigned and developed as an MIT licensed Opensourced product by a bunch of developers and engineers in Bangalore for greater good.
This modules implements a dispatcher for destination addresses. It computes hashes over parts of the request and selects an address from a destination set. The selected address is used then as outbound proxy.
The module can be used as a stateless load balancer, having no guarantee of fair distribution.
Exported Parameters
list_file (string) – Path to the file with destination sets. Default value is “/etc/opensips/dispatcher.list” or “/usr/local/etc/opensips/dispatcher.list”.
setid_col (string) – storing the gateway’s group id. Default value is “setid”.
modparam("dispatcher", "setid_col", "groupid")
destination_col (string) – destination’s sip uri.
modparam("dispatcher", "destination_col", "uri")
flags_col (string) – The column’s name in the database storing the flags for destination uri.
modparam("dispatcher", "flags_col", "dstflags")
force_dst (int) – If set to 1, force overwriting of destination address when that is already set. Default value is “0”.
modparam("dispatcher", "force_dst", 1)
flags (int) – affect dispatcher’s behaviour, defined as a bitmask on an integer value. If flag 1 is set only the username part of the uri will be used when computing an uri based hash. If no flags are set the username, hostname and port will be used The port is used only if different from 5060 (normal sip uri) or 5061 (in the sips case). If flag 2 is set, then the failover support is enabled. The functions exported by the module will store the rest of addresses from the destination set in AVP, and use these AVPs to contact next address when the current-tried fails. Default value is “0”.
modparam("dispatcher", "flags", 3)
use_default (int) – If the parameter is set to 1, the last address in destination set is used as last option to send the message. For example, it is good when wanting to send the call to an anouncement server saying: “the gateways are full, try later”. Default value is “0”.
modparam("dispatcher", "use_default", 1)
dst_avp (str) – The name of the avp which will hold the list with addresses, in the order they have been selected by the chosen algorithm. If use_default is 1, the value of last dst_avp_id is the last address in destination set. The first dst_avp_id is the selected destinations. All the other addresses from the destination set will be added in the avp list to be able to implement serial forking.
modparam("dispatcher", "dst_avp", "$avp(i:271)")
grp_avp (str) – The name of the avp storing the group id of the destination set. Good to have it for later usage or checks. Default is null.
modparam("dispatcher", "grp_avp", "$avp(i:272)")
cnt_avp (str) – The name of the avp storing the number of destination addresses kept in dst_avp avps.
modparam("dispatcher", "cnt_avp", "$avp(i:273)")
hash_pvar (str) – String with PVs used for the hashing algorithm like to do hashing over custom message parts.Default value is “null” – disabled.
ds_ping_method (string) – With this Method you can define, with which method you want to probe the failed gateways. This method is only available, if compiled with the probing of failed gateways enabled. Default value is “OPTIONS”.
modparam("dispatcher", "ds_ping_method", "INFO")
ds_ping_from (string) – With this Method you can define the “From:”-Line for the request, sent to the failed gateways. This method is only available, if compiled with the probing of failed gateways enabled. Default value is “sip:dispatcher@localhost”.
ds_ping_interval (int – With this Method you can define the interval for sending a request to a failed gateway. This parameter is only used, when the TM-Module is loaded. If set to “0”, the pinging of failed requests is disabled.Default value is “10”.
modparam("dispatcher", "ds_ping_interval", 30)
ds_probing_threshhold (int) – If you want to set a gateway into probing mode, you will need a specific number of requests until it will change from “active” to probing. The number of attempts can be set with this parameter. Default value is “3”.
ds_probing_mode (int) – Controls what gateways are tested to see if they are reachable. If set to 0, only the gateways with state PROBING are tested, if set to 1, all gateways are tested. If set to 1 and the response is 407 (timeout), an active gateway is set to PROBING state. Default value is “0”.
modparam("dispatcher", "ds_probing_mode", 1)
options_reply_codes (str) – This parameter must contain a list of SIP reply codes separated by comma. The codes defined here will be considered as valid reply codes for OPTIONS messages used for pinging, apart for 200. Default value is “NULL”.
ds_select_dst(set, alg) – The method selects a destination from addresses set.Meaning of the parameters is as follows: set – the id of the set from where to pick up destination address. It is the first column in destination list file. alg – the algorithm used to select the destination address. “0” – hash over callid “1” – hash over from uri. “2” – hash over to uri. “3” – hash over request-uri. “4” – round-robin (next destination). “5” – hash over authorization-username (Proxy-Authorization or “normal” authorization). If no username is found, round robin is used. “6” – random (using rand()). “7” – hash over the content of PVs string. Note: This works only when the parameter hash_pvar is set. “X” – if the algorithm is not implemented, the first entry in set is chosen.
If the bit 2 in ‘flags’ is set, the rest of the addresses from the destination set is stored in AVP list. You can use ‘ds_next_dst()’ to use next address to achieve serial forking to all possible destinations. This function can be used from REQUEST_ROUTE.
ds_select_dst("1", "0");
ds_select_domain(set, alg) The method selects a destination from addresses set and rewrites the host and port from R-URI. This function can be used from REQUEST_ROUTE.
ds_next_dst() – Takes the next destination address from the AVPs with id ‘dst_avp_id’ and sets the dst_uri (outbound proxy address). This function can be used from FAILURE_ROUTE.
ds_next_domain() – Takes the next destination address from the AVPs with id ‘dst_avp_id’ and sets the domain part of the request uri. This function can be used from FAILURE_ROUTE.
ds_mark_dst() – Mark the last used address from destination set as inactive, in order to be ingnored in the future. In this way it can be implemented an automatic detection of failed gateways. When an address is marked as inactive, it will be ignored by ‘ds_select_dst’ and ‘ds_select_domain’. This function can be used from FAILURE_ROUTE.
ds_mark_dst(“s”) – Mark the last used address from destination set as inactive (“i”/”I”/”0”), active (“a”/”A”/”1”) or probing (“p”/”P”/”2”). With this function, an automatic detection of failed gateways can be implemented. When an address is marked as inactive or probing, it will be ignored by ‘ds_select_dst’ and ‘ds_select_domain’. possible parameters: “i”, “I” or “0” – the last destination should be set to inactive and will be ignored in future requests. “a”, “A” or “1” – the last destination should be set to active. “p”, “P” or “2” – the last destination will be set to probing. Note: You will need to call this function “threshhold”-times, before it will be actually set to probing.This function can be used from FAILURE_ROUTE.
ds_is_from_list() – This function returns true, if the current request comes from a host from the dispatcher-list; otherwise false. This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE and ONREPLY_ROUTE.
ds_is_from_list(“group”) – This function returns true, if the current request comes from a host in the given group of the dispatcher-list; otherwise false.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE and ONREPLY_ROUTE.
Exported MI Functions
ds_set_state – Sets the status for a destination address (can be use to mark the destination as active or inactive).
Parameters: state : state of the destination address
“a”: active
“i”: inactive
“p”: probing group: destination group id address: address of the destination in the group
MI FIFO Command Format:
:ds_set_state:reply_fifo_file state group address empty_line
ds_list – It lists the groups and included destinations.
MI FIFO Command Format:
:ds_list:reply_fifo_file empty_line
ds_reload – reloads the groups and included destinations.
MI DATAGRAM Command Format:
":ds_reload:\n."
Installation and Running
Destination List File – Each destination point must be on one line. First token is the set id, followed by destination address. Optionally, the third field can be flags value (1 – destination inactive, 2 – destination in probing mod — you can do bitwise OR to set both flags). The set id must be an integer value. Destination address must be a valid SIP URI. Empty lines or lines starting with “#” are ignored.
Exmaple of a dispatcher list file
line format
setit(integer) destination(sip uri) flags (integer, optional)
Implementation for a simplified database engine based on text files. It can be used by OpenSIPS DB interface instead of other database module (like MySQL). It keeps everything in memory.
The db_text database system architecture contains
a database is represented by a directory in the local file system. NOTE: when you use db_text in OpenSIPS, the database URL for modules must be the path to the directory where the table-files are located, prefixed by “text://”,
e.g., “text:///var/dbtext/opensips”.
a table is represented by a text file inside database directory.
Internal format of a db_text table
column definition name(type,attr) where types can be int , double , str and attributes be auto , null ,
* each other line is a row with data. The line ends with “\n”.
* the fields are separated by “:”.
* no value between two ‘:’ (or between ‘:’ and start/end of a row) means “null” value.
* next characters must be escaped in strings: “\n”, “\r”, “\t”, “:”.
* 0 — the zero value must be escaped too.
This database interface don’t support the data insertion with default values. All such values specified in the database
template are ignored.
db_mode (integer) – Set caching mode (0 – default) or non-caching mode (1). In caching mode, data is loaded at startup. In non-caching mode, the module check every time a table is requested whether the correspondingfile on disk has changed, and if yes, will re-load table from file.
modparam("db_text", "db_mode", 1)
Exported MI Functions
dbt_dump – Write back to hard drive modified tables.
opensipsctl fifo dbt_dump
dbt_reload – Causes db_text module to reload cached tables from disk. Parameters:
1 db_name (optional) – database name to reload.
2 table_name (optional, but cannot be present without the
db_name parameter) – specific table to reload.
-- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
-- usrloc params --
modparam("usrloc", "db_mode", 2)
modparam("usrloc|auth_db", "db_url", "text:///tmp/opensipsdb")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
route{
// initial sanity checks -- messages with max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if ($ml >= 65535 ) {
sl_send_reply("513", "Message too big");
exit;
};
// we record-route all messages -- to make sure that subsequent messages will go through this proxy;
if (!$rm=="REGISTER") record_route();
<pre><code>// subsequent messages withing a dialog should take the path determined by record-routing
if (loose_route()) {
// mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
exit;
};
if (!is_myself("$rd")) {
// mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
exit;
};
//if the request is for other domain use UsrLoc
if (is_myself("$rd")) {
if ($rm=="REGISTER") {
# digest authentication
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
};
save("location");
exit;
};
lookup("aliases");
if (!is_myself("$rd")) {
append_hf("P-hint: outbound alias\r\n");
route(1);
exit;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);</code></pre>
}
route[1]
{
if (!t_relay()) {
sl_reply_error();
};
}
It is an multi-functional, multi-purpose SIP server especially used in VoIP landscape as standalone SIP server or SBC ( Session Border Controller ) for inbound and outbound traffic by carriers, telecoms backend layers or ITSPs for call routing and trunking solutions. It can be deployed with Class4/5 Platforms, SIP Trunking , hosted or IP PBX setup , existing gateways/ Session Border Controllers, Application Servers, proxy server, Front-End Load Balancers, IMS Platforms, Call Center etc.
Due to its very flexible and customisable routing engine it can be used in number of scenarios such as an SIP proxy or a router and due to its high throughput it is widely recommended as an enterprise grade inbound/outbound proxy server.
SIP to XMPP gateway for presence and IM (bidirectional)
Load-balancer or dispatcher
Inbound/front end for gateways/asterisk
SIP NAT traversal unit
Application server with custom logic
Since Opensips has emerged as a resilent SIP server , it is also used in specific usecases such as – DID ( Direct Inward dialling ) for SIP trunking solutions , – Local Number Portability (LNP) providers, – Canonical Name (CNAME) providers etc
Class 4 routing capabilities in opensips include SIP aliases, Direct Inward Dialing , Speed dial, CPL,vDialplan , dispatcher with various algorithms, prefix-based routing to multiple carriers , failover support, Load balancing , ENUM-based or Geolocation-based routing etc. Some of the Class 5 capabilities in opensips are B2B , call queuing UAC registration , authentication, mangling, White/Black list
SIP SIMPLE features as messaging , Presence , Busy Lamp Field (BLF), Shared Call/Line Appearance (SCA) , Bridged Line Appearance (BLA) , Message Waiting notifications (MWI), XCAP , Resource List Server ( RLS ) , XMPP , SMS gateway (AT and SMPP)
Although opensips has no built-in media capabilities, but it modules for external media engines for Media relaying (RTPProxy, MediaProxy, RTPEngine), Media transcoding (Sangoma D1 cards) , Codec manipulation.
Opensips has majorly 2 parts core and addon-modules.
Opensips Core part is only a proxy stateless SIP server . It contains
SIP transport layer which supports UDP, TCP, TLS and WS for SIP. As per the listener in routing script transport protocols is selected .
SIP factory — the message parser and builder which can be used to add new headers or remove existing ones.
Routing script parser and interpreter for the routing script which loads it to the memory at the startup time. To load a new script server restart is required.
Memory and locking manager for the memory allocation and locking to prevent deadlocks and starvation. Although these arn’t accesible by route scripting, it can be configured at compile time.
Core script functions and variables which can be used in routing scripts in addition to the functions exported by add-on modules.
Interfaces
Events Interface
Used to notify external applications about events triggered internal to OpenSIPS such as
core events – E_CORE_THRESHOLD ,E_CORE_PKG_THRESHOLD , E_CORE_SHM_THRESHOLD , modules events , or even a custom event using raise_event() command
Statistics Interface
Provide insights to statistics of opensips in numerical results which could be used for services like monitoring, load evaluation, realtime integration etc. The statictsics can be of two kinds :
1. counter like – variables that keep counting things that happened in OpenSIPS, like received requests, processed dialogs, failed DB queries, etc
2. computed values – variables that are calculated in realtime, like how much memory is used, the current load, active dialogs, active transactions, etc
These variable would reset form 0 at start sometimes even during runtime.
Binary Internal Interface
Provider communication between individual OpenSIPS instances. Used in cases such as failovers where dialogs needs to persist for service continuity. Hence with this interface one can replicate all the events related to the runtime data (creation / updating / deletion) to a backup OpenSIPS instance.
SQL interface and NoSQL interface
SQL interfaces provides interaction with Sql DB drivers and services such as MySQL, Postgres, Oracle, Berkeley, unixODBC etc , while NoSQL interface provides access to Redis, CouchBase, Cassandra, MongoDB, Memcached, and other databases which are more frequently implemented as external caches.
AAA interface definition
Currently, OpenSIPS supports the RADIUS driver for the AAA interface with upcoming support for DIAMETER.
Management interface
Allows the external applications to trigger predefined commands
Push data like setting a debug level, registering a contact etc
Fetch data like registered users, ongoing calls, get statistics etc
Trigger an internal action as reloading the data, sending a message so on
1. Functional SIP modules
SIP signalling modules such as B2B_ENTITIES , B2B_LOGIC , CALL CENTER ( for Inbound call center system ) , DIALOG , NAT_TRAVERSAL , NATHELPER
OPTIONS , REGISTRAR ,SIGNALING , UAC_REGISTRANT
TM (Transaction/stateful module) , SL (Stateless replier ) , SMS (SIP-to-SMS IM gateway)
A config file opensips.config has 3 main logical parts :
1.global parameters – network listeners, available transport protocols, forking (and number of processes), the logging
2.modules section – the modules that are to be loaded with path to their .so file
3.routing logic – logic for routing sip traffic
Routes
OpenSIPS routing logic uses several types of routes. Each type of route is triggered by a certain event and allows you to process a certain type of message (request or reply).
route
SIP requests routing. The main ‘route’ block identified by ‘route{…}’ or ‘route[0]{…}’ is executed for each SIP request.
To send a reply or forward the request, explicit actions must be called inside the route block. in example below which sends 200 ok reply for each options request.
branch_route
Handles different branches of a SIP request. if the branch is not dropped the branch will be automatically sent out. It is executed only by TM module after it was armed via t_on_branch(“branch_route_index”).
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
or lookup location and discard branches where uri matches ip 1.2.3.4 by using drop()
failure_route
Failed transaction routing block. It contains a set of actions to be taken each transaction that received only negative replies (>=300) for all branches which completes the transaction. The ‘failure_route’ is executed only by TM module after it was armed via t_on_failure(“failure_route_index”).
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
}
onreply_route
Reply routing block. It can be stateful (if bound to a transaction) or stateless (if global reply route).
If the reply is not dropped (only provisional replies can be), it will be injected and processed by the transaction engine. There are three types of onreply routes:
global – catches all replies and uses simple definition ‘onreply_route {…}’ or ‘onreply_route[0] {…}’.
Exmaple for “global” reply route set the whole transaction
route {
seturi("sip:bob@opensips.org"); first branch
append_branch("sip:alice@opensips.org"); second branch
t_on_reply("global");
t_on_branch("1");
t_relay();
}
onreply_route {
xlog("OpenSIPS received a reply from $si\n");
}
onreply_route[global] {
if (t_check_status("1[0-9][0-9]")) {
setflag(1);
log("provisional reply received\n");
if (t_check_status("183"))
drop;
}
}
per request/transaction – it catches all received replies belonging to a certain transaction and uses “t_on_reply()” at request time, in REQUEST ROUTE – named ‘onreply_route[N] {…}’.
per branch – it catches only the replies that belong to a certain branch from a transaction via “t_on_reply()” ) at request time, but in BRANCH ROUTE, when a certain outgoing branch is processed – named ‘onreply_route[N] {…}’.
Certain ‘onreply_route’ blocks can be executed by TM module for special replies. For this, the ‘onreply_route’ must be armed for the SIP requests whose replies should be processed within it, via t_on_reply(“onreply_route_index”).
Exmaple of reply route set for this branch only
branch_route[1] {
if ($rU=="alice")
t_on_reply("alice");
}
onreply_route[alice] {
xlog("received reply on the branch from alice\n");
}
5. error_route
executed automatically on meeting and error such as parsing error in SIP request processing, script assert failure. Performs error handling . The Default action is to discard request. In error_route, the following pseudo-variables are available to get access to error details:
$(err.class) - the class of error (now is '1' for parsing errors)
$(err.level) - severity level for the error
$(err.info) - text describing the error
$(err.rcode) - recommended reply code
$(err.rreason) - recommended reply reason phrase
error_route {
xlog("--- error route class=$(err.class) level=$(err.level)
info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n");
xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n");
sl_send_reply("$err.rcode", "$err.rreason");
exit;
}
local_route
executed automatically as TM created a new request, internally (no UAC side). This is a route intended to be used for message inspection, accounting and for applying last changes on the message headers. Routing and signaling functions are not allowed.
startup_route
Executed only once when OpenSIPS is started and before the processing of SIP messages begins. Used in initilization cases cases such as loading some data in the cache.
startup_route {
avp_db_query("select gwlist where ruleid==1",$avp(i:100));
cache_store("local", "rule1", "$avp(i:100)");
}
timer_route
Route executed periodically at a configured interval of time specified next to the name(in seconds).
timer_route[gw_update, 300] {
avp_db_query("select gwlist where ruleid==1",$avp(i:100));
$shv(i:100) =$avp(i:100);
}
event_route
execute script code when an event is triggered. If no way to handle the event specified, default will be synchronously.
Triggered by the event_route module when an event is raised by the OpenSIPS Event Interface such as event raised by the pike module when it decides an ip should be blocked called E_PIKE_BLOCKED or E_SCRIPT_EVENT etc ( checke events interface for more events)
event_route[E_PIKE_BLOCKED] {
xlog("The E_PIKE_BLOCKED event was raised\n");
}
event_route[E_PIKE_BLOCKED, async] {
xlog("The E_PIKE_BLOCKED event was raised\n");
}
Scripting Language
Opensips scripting provided more advanced controls
1. Core Keywords
Keywords specific to SIP messages which can be used mainly in ‘if’ expressions. af – address family of the received SIP message. It is INET if the message was received over IPv4 or INET6 if the message was received over IPv6.
if(af==INET6) {
log("Message received over IPv6 link\n");
};
dst_ip – IP of the local interface where the SIP message was received.
if(dst_ip==127.0.0.1) {
log("message received on loopback interface\n");
};
dst_port – local port where the SIP packet was received
if(dst_port==5061)
{
log("message was received on port 5061\n");
};
from_uri – reference to the URI of ‘From’ header.
if(is_method("INVITE") &amp;amp;amp;amp;amp;&amp;amp;amp;amp;amp; from_uri=~".*@opensips.org")
{
log("the caller is from opensips.org\n");
};
method – SIP method of the message.
if(method=="REGISTER")
{
log("this SIP request is a REGISTER message\n");
};
msg:len – the size of the message
if(msg:len&amp;amp;amp;amp;gt;2048)
{
sl_send_reply("413", "message too large");
exit;
};
$retcode – value returned by last function executed like $?. If tested after a call of a route, it is the value retuned by that route.
route {
route(1);
if($retcode==1)
{
log("The request is an INVITE\n");
};
}
route[1] {
if(is_method("INVITE"))
return(1);
return(2);
}
proto – transport protocol of the SIP message.
if(proto==UDP)
{
log("SIP message received over UDP\n");
};
status – status code of the reply.
if(status=="200")
{
log("this is a 200 OK reply\n");
};
1.10 src_ip – source IP address
if(src_ip==127.0.0.1)
{
log("the message was sent from localhost!\n");
};
1.11 src_port – source port of the SIP message (from which port the message was sent by previous hop).
if(src_port==5061)
{
log("message sent from port 5061\n");
}
1.12 to_uri – URI from To header.
if(to_uri=~"sip:.+@opensips.org")
{
log("this is a request for opensips.org users\n");
};
1.13 uri – request URI.
if(uri=~"sip:.+@opensips.org")
{
log("this is a request for opensips.org users\n");
};
2. Core Values
Values that can be used in ‘if’ expressions to check against Core Keywords
2.1 INET – IPv4 connection.
2.2 INET6 – IPv6 connection.
2.3 TCP
2.4 UDP
2.5 max_len – test message’s size.
if(msg:len&amp;amp;amp;amp;gt;max_len)
{
sl_send_reply("413", "message too large to be forwarded over UDP without fragmentation");
exit;
}
2.6 myself – reference to the list of local IP addresses, hostnames and aliases that has been set in OpenSIPS configuration file. This lists contain the domains served by OpenSIPS.
if(uri==myself) {
log("the request is for local processing\n");
};
2.7 null – reset the value of a per-script variable or to delete an avp.
$avp(i:12) = null;
$var(x) = null;
3. Core parameters
abort_on_assert – Set to true in order to make OpenSIPS shut down immediately in case a script assert fails.
abort_on_assert = true // default is false
advertised_address – address advertised in Via header and other destination lumps (e.g RR header).
advertised_port – port advertised in Via header and other destination lumps (e.g. RR).
auto_aliases – to control if aliases should be automatically discovered from DNS lookup and added during fixing listening sockets.
auto_aliases=no // default value is no
auto_aliases=0
cfg_file – Returns the name of the corresponding OpenSIPS config file
cfg_line – corresponding line inside the OpenSIPS config file. check_via – Check if the address in top most via of replies is local.
check_via=1 // Default value is 0 (check disabled)
children – Number of worker processes (children) to be created for each UDP or SCTP interface you have defined.
children=16 // Default value is 8 chroot – If set, OpenSIPS will chroot (change root directory) to this valid path in the system value.
chroot=/other/diffroot
debug_mode – This option will automatically force:
staying in foreground set logging level to 4 (debug) set logging to standard error enable core dumping set UDP worker processes to 2 set TCP worker processes to 2 Default value is false/0 (disabled).
db_version_table – name of the table version to be used by the DB API to check the version of the used tables.
db_default_url – default DB URL to be used by modules if no per-module URL is given.
db_max_async_connections – Maximum number of TCP connections opened from a single OpenSIPS worker to each individual SQL backend. Default value is 10. Individual backends are determined from DB URLs as follows: [ scheme, user, pass, host, port, database ]
disable_503_translation – If ‘yes’, OpenSIPS will not translate the received 503 replies into 500 replies . disable_core_dump – By default core dump limits are set to unlimited or a high enough value. Set this config variable to ‘yes’ to disable core dump-ing (will set core limits to 0).
disable_core_dump=yes //Default value is 'no'.
disable_dns_blacklist– DNS resolver, when configured with failover, can automatically store in a temporary blacklist the failed destinations. This will prevent (for a limited period of time) OpenSIPS to send requests to destination known as failed. So, the blacklist can be used as a memory for the DNS resolver.
The temporary blacklist created by DNS resolver is named “dns” and it is by default selected for usage (no need use the use_blacklist()) function. The rules from this list have a life time of 4 minutes – you can change it at compile time, from resolve.c . Can be ‘yes’ or ‘no’. By default the blacklist is disabled (Default value is ‘yes’).
disable_dns_failover – By default DNS-based failover is enabled. Set this config variable to ‘yes’ to disable the DNS-based failover. This is a global option, affecting the core and the modules also.
disable_stateless_fwd – controls the handling of stateless replies:
yes – drop stateless replies if stateless fwd functions (like forward) are not used in script
no – forward stateless replies
dns – controls if the SIP server should attempt to lookup its own domain name in DNS. Default is no.
dns_retr_time – Time in seconds before retrying a dns request. Default value is system specific, depends also on the ‘/etc/resolv.conf’ content (usually 5s).
dns_retr_no – Number of dns retransmissions before giving up.
dns_servers_no – How many dns servers from the ones defined in ‘/etc/resolv.conf’ will be used. Default value is to use all of them.
dns_try_ipv6 – If it is set to ‘yes’ and a DNS lookup fails, it will retry it for ipv6 (AAAA record). Default value is ‘no’.
dns_try_naptr – Disables the NAPTR lookups when doing DNS based routing for SIP requests – if disabled, the DNS lookup will start with SRV lookups. By default it is enabled, value ‘yes’.
dns_use_search_list
dst_blacklist – static (read-only) IP/destination blacklist. These lists can be selected from script (at runtime) to filter the outgoing requests, based on IP, protocol, port, etc.
filter out requests going to ips of my gws
dst_blacklist = gw:{( tcp , 192.168.2.200 , 5060 , “” ),( any , 192.168.2.201 , 0 , “” )}
block requests going to “evil” networks
dst_blacklist = net_filter:{ ( any , 192.168.1.120/255.255.255.0 , 0 , “” )}
block message requests with nasty words
dst_blacklist = msg_filter:{ ( any , 192.168.20.0/255.255.255.0 , 0 , “MESSAGE*ugly_word” )}
block requests not going to a specific subnet
dst_blacklist = net_filter2:{ !( any , 192.268.30.0/255.255.255.0 , 0 , “” )}
Each rule is defined by:
protocol : TCP, UDP, TLS or “any” for anything
port : number or 0 for any
ip/mask
test patter – is a filename like matching (see “man 3 fnmatch”) applied on the outgoing request buffer (first_line+hdrs+body)
enable_asserts – Set to true in order to enable the assert script statement.
enable_asserts = true
event_pkg_threshold – A number representing the percentage threshold above which the E_CORE_PKG_THRESHOLD event is raised, warning about low amount of free private memory. It accepts integer values between 0 and 100. Default value is 0 ( event disabled ).
event_pkg_threshold = 90
event_shm_threshold A number representing the percentage threshold above which the E_CORE_SHM_THRESHOLD event is raised, warning about low amount of free shared memory. It accepts integer values between 0 and 100. Default value is 0 ( event disabled ).
event_shm_threshold = 90
exec_dns_threshold – A number representing the maximum number of microseconds a DNS query is expected to last. Anything above the set number will trigger a warning message to the logging facility. Default value is 0 ( logging disabled ). exec_dns_threshold = 60000
exec_msg_threshold – A number representing the maximum number of microseconds the processing of a SIP msg is expected to last. Anything above the set number will trigger a warning message to the logging facility. Aside from the message and the processing time, the most time consuming function calls from the script will also be logged.
import_file – Same as include_file but will not throw an error if file is not found.
import_file “proxy_regs.cfg”
listen – Set the network addresses the SIP server should listen to. syntax is protocol:address[:port]
The listen definition may accept several optional parameters for:
configuring an advertised IP and port only for an interface. Syntax “AS 11.22.33.44:5060”
setting a different number of children for this interface only (for UDP, SCTP and HEP_UDP interfaces only). This will override the global “children” parameter. Syntax “use_children 5”
Remember that the above parameters only affect the interface they are configured for; if they are not defined for a given interface, the global values will be used instead.
listen = udp:*
listen = udp:eth1
listen = tcp:eth1:5062
listen = tls:localhost:5061
listen = hep_udp:10.10.10.10:5064
listen = ws:127.0.0.1:5060 use_children 5
listen = sctp:127.0.0.1:5060 as 99.88.44.33:5060 use_children 3
On startup, OpenSIPS reports all the interfaces that it is listening on. The TCP engine processes will be created regardless if you specify only UDP interfaces here.
3.41 log_facility – control the facility for logging in syslog. Default value is LOG_DAEMON.
log_facility=LOG_LOCAL0
3.42 log_level – logging level (how verbose OpenSIPS should be). Higher values make OpenSIPS to print more messages.
log_level=1 — print only important messages (like errors or more critical situations) recommended for running proxy as daemon
log_level=4 — print a lot of debug messages use it only when doing debugging sessions
Actual values are:
-3 – Alert level
-2 – Critical level
-1 – Error level
1 – Warning level
2 – Notice level
3 – Info level
4 – Debug level
The ‘log_level’ parameter is usually used in concordance with ‘log_stderror’ parameter.
Value of ‘log_level’ parameter can also be get and set dynamically using log_level Core MI function or $log_level script variable.
3.43 log_name – Set the id to be printed in syslog. The value must be a string and has effect only when OpenSIPS runs in daemon mode (fork=yes), after daemonize. Default value is argv[0].
log_name=”osips-5070″
3.44 log_stderror – write log messages to standard error. Possible values are: – “yes” – write the messages to standard error – “no” – write the messages to syslog , also the default
max_while_loops – maximum loops that can be done within a “while”. Comes as a protection to avoid infinite loops in config file execution. Default is 100.
max_while_loops=200
maxbuffer – size in bytes not to be exceeded during the auto-probing procedure of discovering the maximum buffer size for receiving UDP messages. Default value is 262144.
maxbuffer=65536
mem-group – Defines a group of modules (by name) to get separate memory statistics.In order for the feature to work you have to run “make generate-mem-stats” and complile with the variable SHM_EXTRA_STATS defined and complile with the variable SHM_SHOW_DEFAULT_GROUP definedwill generate the statistics for the default group
mem_warming – Only relevant when the HP_MALLOC compile flag is enabled. If set to “on”, on each startup, OpenSIPS will attempt to restore the memory fragmentation pattern it had before the stop/restart. Memory warming is useful when dealing with high volumes of traffic (thousands of cps on multi-core machines – the more cores, the more useful), because processes must mutually exclude themselves when chopping up the initial big memory chunk. By performing fragmentation on startup, OpenSIPS will also behave optimally in the first minute(s) after a restart. Fragmentation usually lasts a few seconds (e.g. ~5 seconds on an 8GB shm pool and 2.4Ghz CPU) – traffic will not be processed at all during this period.
mem_warming = on
mem_warming_percentage – How much of OpenSIPS’s memory should be fragmented with the pattern of the previous run, upon a restart.
mem_warming_percentage = 50 //Default value: 75
mem_warming_pattern_file – Default value: “CFG_DIR/mem_warming_pattern”.The memory fragmentation pattern of a previous OpenSIPS run. Used at startup, if mem_warming is enabled.
memdump | mem_dump – Log level to print memory status information (runtime and shutdown). Default: memdump=L_DBG (4)
memlog | mem_lo – Log level to print memory debug info. It has to be less than the value of ‘log_level’ parameter if you want memory info to be logged. Default: memlog=L_DBG (4)
mcast_loopback – If set to ‘yes’, multicast datagram are sent over loopback. Default value is ‘no’.
mcast_loopback=yes
mcast_ttl – Set the value for multicast ttl. Default value is OS specific (usually 1).
mhomed – Set the server to try to locate outbound interface on multihomed host. By default is not (0) – it is rather time consuming.
mhomed=1
mpath – Set the module search path. This can be used to simplify the loadmodule parameter
open_files_limit – If set and bigger than the current open file limit, OpenSIPS will try to increase its open file limit to this number. Note: OpenSIPS must be started as root to be able to increase a limit past the hard limit (which, for open files, is 1024 on most systems).
open_files_limit=2048
poll_method – (deprecated post 2.2) poll method to be used by the I/O internal reactor – by default the best one for the current OS is selected. The available types are: poll, epoll_lt, sigio_rt, select, kqueue, /dev/poll.
Starting with version 2.2, epoll_et is deprecated and if it is used in the script, it will be automatically replaced by epoll_lt.
poll_method=select
port – port the SIP server listens to. The default value for it is 5060.
reply_to_via – If it is set to 1, any local reply is sent to the address advertised in top most Via of the request. Default value is 0 (off).
reply_to_via=0
query_buffer_size -If set to a value greater than 1, inserts to DB will not be flushed one by one. Rows to be inserted will be kept in memory until until they gather up to query_buffer_size rows, and only then they will be flushed to the database.
query_buffer_size=5
query_flush_time – If query_buffer_size is set to a value greater than 1, a timer will trigger once every query_flush_time seconds, ensuring that no row will be kept for too long in memory.
query_flush_time=10
rev_dns – should the SIP server attempt to lookup its own IP address in DNS. If this parameter is set to yes and the IP address is not in DNS a warning is printed on syslog and a “received=” field is added to the via header. Default is no.
server_header – The body of Server header field generated by OpenSIPS when it sends a request as UAS. It defaults to “OpenSIPS (<version> (<arch>/<os>))”.
server_header=”Server: My Company SIP Proxy”
server_signature – control “Server” header in any locally generated message. If it is enabled (default=yes) a header is generated as Server: OpenSIPS (0.9.5 (i386/linux))
shm_hash_split_percentage – Only relevant when the HP_MALLOC compile flag is enabled. It controls how many memory buckets will be optimized. (e.g. setting it to 2% will optimize the first 81 most used buckets as frequency). The default value is 1.
shm_secondary_hash_size – Only relevant when the HP_MALLOC compile flag is enabled. It represents the optimization factor of a single bucket (e.g. setting it to 4 will cause the optimized buckets to be further split into 4). The default value is 8.
sip_warning – Can be 0 or 1. If set to 1 (default value is 0) a ‘Warning’ header is added to each reply generated by OpenSIPS. The header contains several details that help troubleshooting using the network traffic dumps.
sip_warning=0
tcp_children – Number of children processes to be created for reading from TCP connections. If no value is explicitly set, the same number of TCP children as UDP children (see “children” parameter) will be used.
tcp_children=4
tcp_accept_aliases – If enabled, OpenSIPS will enforce RFC 5923 behaviour when detecting an “;alias” Via header field parameter and will reuse any TCP (or TLS, WS, WSS) connection opened for such SIP requests (source IP + Via port + proto) when sending other SIP requests backwards, towards the same (source IP + Via port + proto) pair. Default value 0 (disabled).
tcp_listen_backlog – maximum length for the queue of pending connections for the TCP listeners. Default configured value is 10.
tcp_connect_timeout – Time in milliseconds before an ongoing blocking attempt to connect will be aborted. Default value is 100ms.
tcp_connect_timeout = 5
tcp_connection_lifetime – Lifetime in seconds for TCP sessions. Default value is defined in tcp_conn.h: define DEFAULT_TCP_CONNECTION_LIFETIME 120.
tcp_connection_lifetime = 3600
tcp_max_connections – maximum number of tcp connections. Default is defined in tcp_conn.h: define DEFAULT_TCP_MAX_CONNECTIONS 2048
tcp_max_connections = 4096
tcp_max_msg_time – maximum number of seconds that a SIP message is expected to arrive via TCP. Default value is 4
tcp_max_msg_time = 8
tcp_no_new_conn_bflag -A branch flag to be used as marker to instruct OpenSIPS not to attempt to open a new TCP connection when delivering a request, but only to reuse an existing one (if available). If no existing conn, a generic send error will be returned.
This is intended to be used in NAT scenarios, where makes no sense to open a TCP connection towards a destination behind a NAT (like TCP connection created during registration was lost, so there is no way to contact the device until it re-REGISTER). Also this can be used to detect when a NATed registered user lost his TCP connection, so that opensips can disable his registration as useless.
tcp_no_new_conn_bflag = TCP_NO_CONNECT<br> ...<br> route {<br> ...<br> if (isflagset(DST_NATED) && $proto == "TCP")<br> setbflag(TCP_NO_CONNECT);<br> ...<br> t_relay("0x02"); // no auto error reply<br> $var(retcode) = $rc;<br> if ($var(retcode) == -6) {<br> xlog("unable to send request to destination");<br> send_reply("404", "Not Found");<br> exit;<br> } else if ($var(retcode) < 0) {<br> sl_reply_error();<br> exit;<br> }<br> }
3.77 tcp_threshold – A number representing the maximum number of microseconds sending of a TCP request is expected to last. Anything above the set number will trigger a warning message to the logging facility. Default value is 0 ( logging disabled ).
tcp_threshold = 60000
tcp_keepalive – Enable or disable TCP keepalive (OS level). Enabled by default.
tcp_keepalive = 1
tcp_keepcount -Number of keepalives to send before closing the connection (Linux only).
Default value: 0 (not set). Setting tcp_keepcount to any value will enable tcp_keepalive.
tcp_keepcount = 5
tcp_keepidle – Amount of time before OpenSIPS will start to send keepalives if the connection is idle (Linux only). Default value: 0 (not set)
tcp_keepidle = 30
tcp_keepinterval – Interval between keepalive probes, if the previous one failed (Linux only).Default value: 0 (not set). Setting tcp_keepinterval to any value will enable tcp_keepalive.
tcp_keepinterval = 10
tls_ca_list
tls_certificate
tls_ciphers_list
tls_domain
tls_handshake_timeout
tls_log
tls_method
tls_port_no
tls_private_key
tls_require_certificate
tls_send_timeout
tls_verify
tos – TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP).
user_agent_header – The body of User-Agent header field generated by OpenSIPS when it sends a request as UAC. It defaults to “OpenSIPS (<version> (<arch>/<os>))”.
user_agent_header=”User-Agent: My Company SIP Proxy”
wdir – working directory used by OpenSIPS at runtime.
wdir="/usr/local/opensips"
xlog_buf_size – Default value: 4096
Size of the buffer used to print a single line on the chosen logging facility of OpenSIPS. If the buffer is too small, an overflow error will be printed, and the concerned line will be skipped.
xlog_buf_size = 8388608 #given in bytes
xlog_force_color
xlog_default_level -Default value for the logging level of the xlog core function, when the log_level parameter is omitted.
OpenSIPS routing logic uses several types of routes. Each type of route is triggered by a certain event and allows you to process a certain type of message (request or reply).
route SIP requests routing. The main ‘route’ block identified by ‘route{…}’ or ‘route[0]{…}’ is executed for each SIP request. To send a reply or forward the request, explicit actions must be called inside the route block. in example below which sends 200 ok reply for each options request.
branch_route Handles different branches of a SIP request. if the branch is not dropped the branch will be automatically sent out. It is executed only by TM module after it was armed via t_on_branch(“branch_route_index”).
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
or lookup location and discard branches where uri matches ip 1.2.3.4 by using drop()
failure_route Failed transaction routing block. It contains a set of actions to be taken each transaction that received only negative replies (>=300) for all branches which completes the transaction. The ‘failure_route’ is executed only by TM module after it was armed via t_on_failure(“failure_route_index”).
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
}
onreply_route Reply routing block. It can be stateful (if bound to a transaction) or stateless (if global reply route). If the reply is not dropped (only provisional replies can be), it will be injected and processed by the transaction engine. There are three types of onreply routes:
global – catches all replies and uses simple definition ‘onreply_route {…}’ or ‘onreply_route[0] {…}’.
Exmaple for “global” reply route set the whole transaction
route {
seturi("sip:bob@opensips.org"); first branch
append_branch("sip:alice@opensips.org"); second branch
t_on_reply("global");
t_on_branch("1");
t_relay();
}
onreply_route {
xlog("OpenSIPS received a reply from $si\n");
}
onreply_route[global] {
if (t_check_status("1[0-9][0-9]")) {
setflag(1);
log("provisional reply received\n");
if (t_check_status("183"))
drop;
}
}
per request/transaction – it catches all received replies belonging to a certain transaction and uses “t_on_reply()” at request time, in REQUEST ROUTE – named ‘onreply_route[N] {…}’.
per branch – it catches only the replies that belong to a certain branch from a transaction via “t_on_reply()” ) at request time, but in BRANCH ROUTE, when a certain outgoing branch is processed – named ‘onreply_route[N] {…}’.
Certain ‘onreply_route’ blocks can be executed by TM module for special replies. For this, the ‘onreply_route’ must be armed for the SIP requests whose replies should be processed within it, via t_on_reply(“onreply_route_index”).
Exmaple of reply route set for this branch only
branch_route[1] {
if ($rU=="alice")
t_on_reply("alice");
}
onreply_route[alice] {
xlog("received reply on the branch from alice\n");
}
error_route
executed automatically on meeting and error such as parsing error in SIP request processing, script assert failure. Performs error handling . The Default action is to discard request. In error_route, the following pseudo-variables are available to get access to error details:
$(err.class) - the class of error (now is '1' for parsing errors)
$(err.level) - severity level for the error
$(err.info) - text describing the error
$(err.rcode) - recommended reply code
$(err.rreason) - recommended reply reason phrase
error_route {
xlog("--- error route class=$(err.class) level=$(err.level)
info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n");
xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n");
sl_send_reply("$err.rcode", "$err.rreason");
exit;
}
local_route
executed automatically as TM created a new request, internally (no UAC side). This is a route intended to be used for message inspection, accounting and for applying last changes on the message headers. Routing and signaling functions are not allowed.
startup_route Executed only once when OpenSIPS is started and before the processing of SIP messages begins. Used in initilization cases cases such as loading some data in the cache.
startup_route {
avp_db_query("select gwlist where ruleid==1",$avp(i:100));
cache_store("local", "rule1", "$avp(i:100)");
}
timer_route Route executed periodically at a configured interval of time specified next to the name(in seconds).
timer_route[gw_update, 300] {
avp_db_query("select gwlist where ruleid==1",$avp(i:100));
$shv(i:100) =$avp(i:100);
}
event_route execute script code when an event is triggered. If no way to handle the event specified, default will be synchronously. Triggered by the event_route module when an event is raised by the OpenSIPS Event Interface such as event raised by the pike module when it decides an ip should be blocked called E_PIKE_BLOCKED or E_SCRIPT_EVENT etc ( checke events interface for more events)
event_route[E_PIKE_BLOCKED] {
xlog("The E_PIKE_BLOCKED event was raised\n");
}
event_route[E_PIKE_BLOCKED, async] {
xlog("The E_PIKE_BLOCKED event was raised\n");
}
Metrics for monitoring a VOIP call can be obtained from any node in media path of the call flow . Essentially used for analysis via calculation and aggregation , and sometimes used for realtime performance tracking and rectification too .
Rating Factor (R-Factor) and Mean Opinion Score (MOS) are two commonly-used measurements of overall VoIP call quality.
R-Factor: A value derived from metrics such as latency, jitter, and packet loss per ITU‑T Recommendation G.107. It assess the quality-of-experience for VoIP calls on your network. Typical scores range from 50 (bad) to 90 (excellent).
R factor of 90 , Mos is 4.3 ( Excellent )
R factor 50 , Mos is 2.6 ( Bad)
MOS: It is derived from the R-Factor per ITU‑T Recommendation G.10 which measures VoIP call quality. PacketShaper measures MOS using a scale of 10-50. To convert to a standard MOS score (which uses a scale of 1-5), divide the PacketShaper MOS value by 10.
ITU ? The International Telecommunication Union is the United Nations specialised agency in the field of telecommunications, information and communication technologies (ICTs).
ITU-T ? TU Telecommunication Standardisation Sector is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardising telecommunications on a worldwide basis.
MOS ( Mean Opinion Score )
MOS is terminology for audio, video and audiovisual quality expressions as per ITU-T P.800.1. It refers to listening, talking or conversational quality, whether they originate from subjective or objective models.
It provides provisions for identifiers regarding the audio bandwidth, the type of interface (electrical or acoustical) and the video resolution too , such as
MOS-AVQE for audiovisual quality;
MOS-CQE is for estimated conversational quality;
MOS-LQE for listening quality;
MOS-TQE is used for talking quality;
MOS-VQE depicts video quality;
For Audio Signal Speech Quality/ AV
– N denotes audio signals upto narrow-band (300-3400 Hz)
– W is for audio signals upto wideband (50-7000 Hz)
– S for upto super-wideband (20-14000 Hz)
– F is obtained for fullband (10-20000 Hz)
For Listening quality LQO
electrical measurement
performed at electrical interfaces only. In order to predict the listening quality as perceived by the user, assumptions for the terminals are made in terms of intermediate reference system (IRS) or corrected IRS frequency response. A sealed condition between the handset receiver and the user’s ear is assumed.
acoustical measurement
performed at acoustical interfaces. In order to predict the listening quality as perceived by the user, this measurement includes the actual telephone set products provided by the manufacturer or vendor. In combination with the choice of the acoustical receiver in the laboratory test , there will be a more or less leaky condition between the handset’s receiver and the artificial ear.
Conversational Quality / CQ
Arithmetic mean value of subjective judgments on a 5-point ACR quality scale, is calculated. Talking Quality / TQ
This describes the quality of a telephone call as it is perceived by the talking party only. Factors affecting TQ include echo signal , background noise , double talk etc. It is calculated based on the arithmetic mean value of judgments on a 5-point ACR quality scale.
Video Quality / VQ
To account for differentiation in perceived quality for mobile and fixed devices and to allow for proper handling of different use-cases as
– M for mobile screen such as a smartphone or tablet (approximately 25 cm or less)
– T for PC/TV monitors
It is calculated based on the arithmetic mean value of subjective judgments, typically on a 5-point quality scale
Audio Visual Quality / AVQ
Refers to quality of audio visual stream under corresponding networking conditions. It is also calculated based on the arithmetic mean value of judgments on a 5-point ACR quality scale.
Other parameters also contributing to VoIP metric Analysis
Latency
It is the time required for packets to travel from one end to another, in milliseconds. If the sum of measured latency is 800 ms and the number of latency samples is 20, then the average latency is 40 ms. Header of the RTP packets carry timestamps which later can also be used to calculate round-trip time.
Packet Loss
packet loss percentage performed per RFC 3550 using RTP header sequence numbers.
Jitter
The variation in the delay of received packets in a flow, measured by comparing the interval when RTP packets were sent to the interval at which they were received.
For instance, if packet #1 and packet #2 leave 30 milliseconds apart and arrive 50 milliseconds apart, then the jitter is 20 milliseconds.
Methods for objective and subjective assessment of speech and video quality.
Mapping R-value to calculate MOS
To map MOS from R value using above defined metrics , a standard formula is used. First the latency and jitter are added and defined value for computation time is also added , resulting in effective latency
This article focuses on setting up sipwise rtpegine to proxy rtp traffic from kamailio app server. This is an updated version of the the old article .
RTPengine is a proxy for RTP traffic and other UDP based media traffic over either IPv4 or IPv6. It can even bridge between diff IP networks and interfaces . It can do TOS/QoS field setting. It is Multi-threaded , can advertise different addresses for operation behind NAT.
It bears in-kernel packet forwarding for low-latency and low-CPU performance .
When used with kamailio RTP engine module it adds more features . I wrote an article covering all relevant and important kamailio modules earlier including RTPProxy and RTP engine ;https://telecom.altanai.com/2014/11/18/kamailio-modules/.
Full SDP parsing and rewriting
Supports non-standard RTCP ports (RFC 3605)
ICE (RFC 5245):
Bridging between ICE-enabled and ICE-unaware user agents
Optionally acting only as additional ICE relay/candidate
Optionally forcing relay of media streams by removing other ICE candidates
SRTP (RFC 3711):
Support for SDES (RFC 4568) and DTLS-SRTP (RFC 5764)
AES-CM and AES-F8 ciphers, both in userspace and in kernel
HMAC-SHA1 packet authentication
Bridging between RTP and SRTP user agents
RTCP profile with feedback extensions (RTP/AVPF, RFC 4585 and 5124)
Arbitrary bridging between any of the supported RTP profiles (RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF)
RTP/RTCP multiplexing (RFC 5761) and demultiplexing
Breaking of BUNDLE’d media streams (draft-ietf-mmusic-sdp-bundle-negotiation)
Recording of media streams, decrypted if possible
Transcoding and repacketization
Playback of pre-recorded streams/announcements
Sipwise NGCP RTP Engine Source Code
There are 3 parts of the source structure in sipwise NGCP ( Next Generation communication Platform) rtpengine :
1.daemon
The userspace daemon and workhorse, minimum requirement for anything to work. Running make will compile the binary, which will be called rtpengine.
Required packages including their development headers are required to compile the daemon:
pkg-config
GLib including GThread and GLib-JSON version 2.x
zlib
OpenSSL
PCRE library
XMLRPC-C version 1.16.08 or higher
hiredis library
gperf
libcurl version 3.x or 4.x
libevent version 2.x
libpcap
libsystemd
MySQL or MariaDB client library (optional for media playback and call recording daemon)
libiptc library for iptables management (optional)
ffmpeg codec libraries for transcoding (optional) such as libavcodec, libavfilter, libswresample
bcg729 for full G.729 transcoding support (optional)
options for make – with_iptables_option , with_transcoding
with_transcoding=no make
2.iptables-extension
Required for in-kernel packet forwarding. With the iptables development headers installed, issuing make will compile the plugin for iptables and ip6tables. The file will be called libxt_RTPENGINE.so and needs to be copied into the xtables module directory. The location of this directory can be determined through pkg-config xtables –variable=xtlibdir on newer systems, and/or is usually either /lib/xtables/ or /usr/lib/x86_64-linux-gnu/xtables/.
3.kernel-module
Required for in-kernel packet forwarding. Compilation of the kernel module requires the kernel development headers to be installed in/lib/modules/$VERSION/build/, where $VERSION is the output of the command uname -r.
Successful compilation of the module will produce the file xt_RTPENGINE.ko. The module can be inserted into the running kernel manually through insmod xt_RTPENGINE.ko
It is recommended to copy the module into /lib/modules/$VERSION/updates/, followed by running depmod -a.
After this, the module can be loaded by issuing modprobe xt_RTPENGINE.
vi debian/control
change from default-libmysqlclient-dev to libmysqlclient-dev, change from libiptcdata-dev to libiptc-dev and install the alternatives such as
apt install libmysqlclient-dev libiptcdata-dev
Generated deb files should be outside the rtpegine home folder
To avoid the overhead involved in processing each individual RTP packet in userspace-only operation, especially as RTP traffic consists of many small packets at high rates, rtpengine provides a kernel module to offload the bulk of the packet forwarding duties from user space to kernel space. This also results in increasing the number of concurrent calls as CPU usage decreases.In-kernel packet forwarding is implemented as an iptables module (x_tables) and has 2 parts – xt_RTPENGINE and plugin to the iptables and ip6tables command-line utilities
Sequence of events for a newly established media stream is then:
Kamailio as SIP proxy controls rtpengine and signals it about a newly established call.
Rtpengine daemon allocates local UDP ports and sets up preliminary forward rules based on the info received from the SIP proxy.
An RTP packet is received on the local port.
It traverses the iptables chains and gets passed to the xt_RTPENGINE module.
The module doesn’t recognize it as belonging to an established stream and thus ignores it.
The packet continues normal processing and eventually ends up in the daemon’s receive queue.
The daemon reads it, processes it and forwards it. It also updates some internal data.
This userspace-only processing and forwarding continues for a little while, during which time information about additional streams and/or endpoints may be obtained from the SIP proxy.
After a few seconds, when the daemon is satisfied with what it has learned about the media endpoints, it pushes the forwarding rules to the kernel.
From this moment on, the kernel module will recognize incoming packets belonging to those streams and will forward them on its own. It will stop those packets from traversing the network stacks any further, so the daemon will not see them any more on its receive queues.
In-kernel forwarding is allowed to cease to work at any given time, either accidentally (e.g. by removal of the iptablesrule) or deliberatly (the daemon will do so in case of a re-invite), in which case forwarding falls back to userspace-only operation.
The Kernel Module
The kernel module supports multiple forwarding tables, identified through their ID number , bydefault 0 to 63
Each running instance of the rtpengine daemon controls one such table. To load use
modprobe xt_RTPENGINE and to unload rmmod xt_RTPENGINE,. With the module loaded, a new directory will appear in /proc/, namely /proc/rtpengine/ , containing pseudo-files, control ( to create and delete forwarding tables) and list ( list of currently active forwarding tables)
To manually create a forwarding table with ID 33, the following command can be used:
echo ‘add 43’ > /proc/rtpengine/control
The iptables module
In order for the kernel module to be able to actually forward packets, an iptables rule must be set up to send packets into the module. Each such rule is associated with one forwarding table. In the simplest case, for forwarding table 33, this can be done through:
iptables -I INPUT -p udp -j RTPENGINE –id 33
To restrict the rules to the UDP port range used by rtpengine, e.g. by supplying a parameter like –dport 30000:40000. If the kernel module receives a packet that it doesn’t recognize as belonging to an active media stream, it will simply ignore it and hand it back to the network stack for normal processing.
A typical start-up sequence including in-kernel forwarding might look like this:
To run multiple instances of rtpengine on the same machine run multiple instances of the daemon using different command-line options ( local addresses and listening ports), together with multiple different kernel forwarding tables.
For example, if one local network interface has address 10.64.73.31 and another has address 192.168.65.73, then the start-up sequence might look like this:
With this setup, the SIP proxy can choose which instance of rtpengine to talk to and thus which local interface to use by sending its control messages to either port 2223 or port 2224.
Transcoding
Currently transcoding is supported for audio streams. Can we turned off with with_transcoding=no option in makeFile
Normally rtpengine leaves codec negotiation up to the clients involved in the call and does not interfere. In this case, if the clients fail to agree on a codec, the call will fail.
transcoding options in the ng control protocol, transcode or ptime . If a codec is requested via the transcode option that was not originally offered, transcoding will be engaged for that call. With transcoding active for a call, all unsupported codecs will be removed from the SDP.
Transcoding happens in userspace only, so in-kernel packet forwarding will not be available for transcoded codecs. Codecs that are supported by both sides will simply be passed through transparently (unless repacketization is active). In-kernel packet forwarding will still be available for these codecs.
codecs supported by rtpengine can be shown with –codecs options
rtpengine –codecs
PCMA: fully supported
PCMU: fully supported
G723: fully supported
G722: fully supported
QCELP: supported for decoding only
G729: supported for decoding only
speex: fully supported
GSM: fully supported
iLBC: not supported
opus: fully supported
vorbis: codec supported but lacks RTP definition
ac3: codec supported but lacks RTP definition
eac3: codec supported but lacks RTP definition
ATRAC3: supported for decoding only
ATRAC-X: supported for decoding only
AMR: supported for decoding only
AMR-WB: supported for decoding only
PCM-S16LE: codec supported but lacks RTP definition
PCM-U8: codec supported but lacks RTP definition
MP3: codec supported but lacks RTP definition
ng Control Protocol
advanced control protocol to pass SDP body from the SIP proxy to the rtpengine daemon, has the body rewritten in the daemon, and then pas back to the SIP proxy to embed into the SIP message. It is based on the bencode standard and runs over UDP transport.
Each message passed between the SIP proxy and the media proxy contains of two parts: message cookie ( to match requests to responses, and retransmission detection) and bencoded dictionary
The dictionary of each request must contain at least one key called command and corresponding value must be a string and determines the type of message. Currently the following commands are defined:
ping
offer
answer
delete
query
start recording
stop recording
block DTMF
unblock DTMF
block media
unblock media
start forwarding
stop forwarding
play media
stop media
The response dictionary must contain at least one key called result. The value can be either ok (optional key warning) or error( to be accompanied by error-reason ). For the ping command, the additional value pong is allowed.
SIPp is an opensource (GNU GPL license) performance testing tool for the SIP protocol and is widely used for Quality assurabce of callflows in voip applications for UAC / UASs cenarios. It can emulate functioing of a sip phone such as REGISTER , establishes and releases multiple calls with
the INVITE and BYE methods , send other SIP requests and wait for reponsesbased on dafult of custom xml scenario files .
Plus factor is the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics),
periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management,
regular expressions and variables in scenario files, and dynamically adjustable call rates.
It is widley used as aperformnace and load testing tool since it can test SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, and SIP PBXes and can also emulate thousands of user agents calling your SIP system.
Installation
Pre-requisites to compile SIPp are:
– C++ Compiler
– curses or ncurses library
– For TLS support: OpenSSL >= 0.9.8
– For pcap play support: libpcap and libnet
– For SCTP support: lksctp-tools
– For distributed pauses: Gnu Scientific Libraries
Download , extract and build with options like
tar -xvzf sipp-xxx.tar.gz
cd sipp
./configure --with-sctp --with-pcap --with-openssl
make
Verify installation Run sipp with embedded server (uas) scenario:
sipp -sn uas
On the same host, run sipp with embedded client (uac) scenario:
sipp -sn uac 127.0.0.1 -trace_msg -trace_err
output for server
# sipp -sn uas
------------------------------ Scenario Screen -------- [1-9]: Change Screen --
Port Total-time Total-calls Transport
5060 32.95 s 61 UDP
0 new calls during 0.874 s period 1 ms scheduler resolution
19 calls Peak was 41 calls, after 28 s
0 Running, 63 Paused, 12 Woken up
0 dead call msg (discarded)
3 open sockets
———————————————– 2019-02-04 13:08:13.940159
UDP message sent (371 bytes):
ACK sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-5
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 1 ACK
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
~ RTP
———————————————– 2019-02-04 13:08:13.941658
UDP message sent (371 bytes):
BYE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
———————————————– 2019-02-04 13:08:13.952888
UDP message received [313] bytes :
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact:
Content-Length: 0
Time
---------------------------- Repartition Screen ------- [1-9]: Change Screen --
Average Response Time Repartition 1
0 ms <= n < 10 ms : 293
10 ms <= n < 20 ms : 9
20 ms <= n < 30 ms : 0
30 ms <= n < 40 ms : 0
40 ms <= n < 50 ms : 0
50 ms <= n < 100 ms : 0
100 ms <= n < 150 ms : 0
150 ms <= n < 200 ms : 0
n >= 200 ms : 0
Average Call Length Repartition
0 ms <= n < 10 ms : 0
10 ms <= n < 50 ms : 0
50 ms <= n < 100 ms : 0
100 ms <= n < 500 ms : 0
500 ms <= n < 1000 ms : 0
1000 ms <= n < 5000 ms : 262
5000 ms <= n < 10000 ms : 0
n >= 10000 ms : 0
------------------------------ Sipp Server Mode -------------------------------
3 new calls during 0.286 s period 1 ms scheduler resolution
0 calls (limit 30) Peak was 25 calls, after 10 s
0 Running, 101 Paused, 7 Woken up
0 dead call msg (discarded) 0 out-of-call msg (discarded)
3 open sockets
----------------------------- Statistics Screen ------- [1-9]: Change Screen --
Start Time | 2019-02-04 13:08:03.908208 1549265883.908208
Last Reset Time | 2019-02-04 13:08:20.954289 1549265900.954289
Current Time | 2019-02-04 13:08:21.241152 1549265901.241152
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:286000 | 00:00:17:332000
Call Rate
Tracings
———————————————– 2019-02-04 13:08:13.934840
UDP message received [527] bytes :
INVITE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-0
From: sipp ;tag=52422SIPpTag001
To: service
Call-ID: 1-52422@192.x.x.x
CSeq: 1 INVITE
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 135
———————————————– 2019-02-04 13:08:13.948679
UDP message received [371] bytes :
ACK sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-5
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 1 ACK
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
~ RTP
———————————————– 2019-02-04 13:08:13.949168
UDP message received [371] bytes :
BYE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
———————————————– 2019-02-04 13:08:13.949245
UDP message sent (313 bytes):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact:
Content-Length: 0
time
---------------------------- Repartition Screen ------- [1-9]: Change Screen --
Average Response Time Repartition 1
0 ms <= n < 10 ms : 657
10 ms <= n < 20 ms : 20
20 ms <= n < 30 ms : 0
30 ms <= n < 40 ms : 0
40 ms <= n < 50 ms : 0
50 ms <= n < 100 ms : 0
100 ms <= n < 150 ms : 0
150 ms <= n < 200 ms : 0
n >= 200 ms : 0
Average Call Length Repartition
0 ms <= n < 10 ms : 649
10 ms <= n < 50 ms : 28
50 ms <= n < 100 ms : 0
100 ms <= n < 500 ms : 0
500 ms <= n < 1000 ms : 0
1000 ms <= n < 5000 ms : 0
5000 ms <= n < 10000 ms : 0
n >= 10000 ms : 0
------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic -----
Last Error: Overload warning: the major watchdog timer 3000ms has been t…
Controlling sipp
[+|-|*|/]: Adjust rate
p Pause traffic
q Quit SIPp (after all calls complete, enter a second time to quit immediately)
Q Quit SIPp immediately s Dump screens to the log file (if -trace_screen is passed)
Blockchain is essentially a decentralized algorithm for distributed storage and processing , using a non immutable data structures and securing them with signatures and keys . These sequential chain of records called blocks , can contains almost anything from timestamped transactions , metadata , contracts , files etc just as long as they are chained using hash pointers to previous blocks .
what is a hash ?
function f (x) = y , takes an i/o and give a determined o/p . example heaxadecimeal output of my name , md5(altanai bisht) = 2b9e76d57842ebafaf19fd33bb3573a3.
These are irreversible ie one cant find the i/p from o/p . For this u need to try every combination using brute force. Hence these are generally used for cross verification without revealing the information itself .
Who are miners ?
-tbd –
Application of block chain :
Market analysts and industry specialist have said that block-chain is a revolutionizing technology which will create a decentralized network for not just currency exchange but also many other aspects such as double spent problem , universal identities , document management etc . Example : Bitcoin protocol , which contains a full record of every transaction ever executed with the currency at any time in past. It is also a solution to problem like black – money , double spending , tax evasions etc. Other areas include:
Decentralizing document keeping such as government records , digital assets , equity information , medical and health records etc . The system also provide data ownership and Intellectual property protection .
Fintech as AML( Anti money laundering) , eKYC ( Know Your customer ) , epay , loans, stock trading .
Smart contracts such as in ethereum . Allows to keep program code that would execute on an event.
Shared economy for a p2p payment system .
Crowdfunding , works on paradigm of token owner’s voting and cooperation in decisions for crowd-sourced venture capital funds .
Micro payments / fractional concurrency for small amounts suits power selling and buying such as on solar renewable power micro grid
Since a block chain is a ledger of facts shared across many peer nodes , all communication and inter node transaction uses the power of crypto to authenticate each other and validate each others requests from the genesis block .
what is a genesis block ?
First block of blockchain which needs to be hard-coded into software . It is the only block which does not reference a previous block .
As any peer wants to add a fact to the ledger , a consensus needs to be obtained from the network. This way of network agreement ensures that fraudulent behavior is prevented .
There is only one path from top block on chain to genesis root , however there can many forks upwards from genesis block . It is so because blocks may be created within a short span of time or be under processing . One of the two block will be added to main chain and other will be orphaned or added to pool of queued transactions or even be lost.
Steps to Programming a simple block-chain application :
Lets assume we are creating a block chain for call records.
Structure of a block which is an object which typically looks like
Blocks have an index , timestamp , transactions ( in our case call status such as outgoing or incoming calls ) and the hash link of previous block , which enables the chain formation ,
Create a class , blockchain , for member function and variables. Create functions as :
init() : create a new chain and transaction object
createNewTranscation( ) : this creates the information which needs to be fed into the next mined block and returns the index of the new block which the transaction will be added to .
createNewBlock() : at first we need to create a genesis block
fetchLastBlock() ,
boolean isBlockValid ( newBlock , oldBlock) – checks if the oldblocks index is sequentially aligned with new block and whether old blocks hash is equal to new blocks previous hash . Also calculates whether hash of new block is actually same as the supplied hash value in new block ( give below) .
hashBlock( block ) – to create the hashes we need to add in block. Basically a SHA 256 hash of concatenated arguments as index, timestamp, message , previous hash and a nonce . example pseudo code :
Consensus Algorithms
All block-chains a\re deterministic state machines and transactions act upon them . Consensus filters out the invalid ones and reaches on agreement with valid ones.
DPOS (Delegated Proof of Stake)
A consensus algorithm used for electing producers and scheduling them in a fair and democratic way . It works on the simple principle that longest chain wins therefore incases of multiple forks or network disruption also , if an honest peer finds out a valid strictly longer chain , it will switch from its current fork to the longer chain. We assume that in all conditions , no other chain forked can be longer if 2/3 of producers are honest as 2/3 + 1 confirmations are required .
In crypto we trust !
Block chain is primarily 3 things : p2p network, public key cryptography and distributed consensus .
The security and accountability of such a system is managed via mass surveillance of transactions and cryptographic evidence. Ensures that blocks are always in chronological order since meddling with the blocks will change the hash for preceding blocks
Verification of block uses ECDSA ( Elliptic Curve Digital Signature Algorithm ) to ensure that tokens are spend by their rightful owners only.
An ellipsis is a derived from the second degree equation like ax^2 + bcy + cy^2 + dx + ey +f =0 . Depending on attributes this could be hyperbola , parabola or even a circle . However elliptic curve cryptography uses a third degree equation from either a pseudo -random curve ( such as over prime fields y^2=x^3+ax+b or binary fields y^2 + xy = x^3 + ax^2 + b ) or a special curve .
what is ECDSA ?
There are 2 types of auth schemes : Symmetric , relying on shared secret key and Asymmetric relying on private public keys . ECDSA is a asymmetric authentication scheme where in addition to sender and receiver , even 3rd party systems can be authenticated . In this the sender uses his private key to sign the message and receiver uses the senders public key to verify the message’s signature .
Difficulty
While publishing a block with pending facts to be appended to a chain , the owner sends it to other nodes for confirmation on its validity. Once its approved , other nodes called miners add it to their copy of chains. However the new block has to be published after fixed time interval for fraud prevention ( example : bitcoin blocks are published every 10 mins on avg ) . This duration is dynamically recalculated as the network miners grow or shrink . A difficulty is a number metric that represents how difficult is it to find a hash for given target.
To force increase time for calculating the matching hash , difficulty is increased for miners work harder and take longer to earn the block reward .
While in case of less miner participation , the block difficulty level is made lower
SIP is the most popular signalling protocol in VOIP ecosystem. It is most suited to a caller-callee scenario , yet however supporting scalable conferences on VOIP is a market demand. It is desired that SIP must for multimedia stream but also provide conference control for building communication and collaboration apps for new and customisable solutions.
Role of SIP in conference involves
initiating confs
inviting participants
enabling them to join conf
leave conf
terminate conf
expel participants
configure media flow
control activities in conf
Centralised vs Mesh signalling for Multi participant conf
In a Centralised signalling model , all communication flows via a centralised control point
In a decentralised or mesh signalling structure , participants can communicate p2p
Unicast vs Multicast Media Distribution
Decentralised Media , Multi unicast streaming
Decentralised media , Multicast
Centralised Media / MCU
Conference types
1. Bridge
Centralised entity to book conf , start conf , leave conf . Therefore single point of failure potentially .
To create conf : conf created on a bridge URL , bridge registers on SIP Server, participants join the conf on the bridge using INVITES
To stop conf : either participant can Leave with BYE or conf can terminate by sending BYE to all
2. Endpoints as Mixer
Endpoints handle stream , decentralised media , therefore adhoc suited
mixer UAs cannot leave untill conf finishes
3. Mesh
coplex and more processing power on each UA required
no single point of failure but endpoints have to handle NATIng
GStreamer ( LGPL )ia a media handling library written in C for applicatioan such as streaming , recording, playback , mixing and editing attributes etc. Even enhnaced applicaiosn such as tsrancoding , media ormat conversion , streaming servers for embeeded devices ( read more about Gstreamer in RPi in my srticle here).
It encompases various codecs, filters and is modular with plugins developement to enhance its capabilities. Media Streaming application developers use it as part of their framework at either the broadcaster’s end or as media player.
gstreamer-bad-video-1.0GStreamer bad video library – Bad video library for GStreamer elementsgstreamer-tag-1.0 GStreamer Tag Library – Tag base classes and helper functionsgstreamer-bad-base-1.0 GStreamer bad base classes – Bad base classes for GStreamer elements
gstreamer-audio-1.0 GStreamer Audio library – Audio helper functions and base classes
gstreamer-plugins-bad-1.0 GStreamer Bad Plugin libraries – Streaming media framework, bad plugins libraries
gstreamer-rtsp-server-1.0 gst-rtsp-server – GStreamer based RTSP server
At the time of writing this article Gstreamer an much early version in 1.X , which was newer than its then stable version 0.x. Since then the library has updated many fold. summarising release highlights for major versions as the blog was updated over time .
Gstreamer 1.8.0 – 24 March 2016
Features Hardware-accelerated zero-copy video decoding on Android
New video capture source for Android using the android.hardware.Camera API
Windows Media reverse playback support (ASF/WMV/WMA)
tracing system provides support for more sophisticated debugging tools
high-level GstPlayer playback convenience API
Initial support for the new Vulkan API
Improved Opus audio codec support: Support for more than two channels; MPEG-TS demuxer/muxer can handle Opus; sample-accurate encoding/decoding/transmuxing with Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container; new codec utility functions for Opus header and caps handling in pbutils library. The Opus encoder/decoder elements were also moved to gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good.
Asset proxy support in the GStreamer Editing Services
GStreamer 1.16.0 – 19 April 2019.
GStreamer WebRTC stack gained support for data channels for peer-to-peer communication based on SCTP, BUNDLE support, as well as support for multiple TURN servers.
AV1 video codec support for Matroska and QuickTime/MP4 containers and more configuration options and supported input formats for the AOMedia AV1 encoder
Closed Captions and other Ancillary Data in video
planar (non-interleaved) raw audio
GstVideoAggregator, compositor and OpenGL mixer elements are now in -base
New alternate fields interlace mode where each buffer carries a single field
WebM and Matroska ContentEncryption support in the Matroska demuxer
new WebKit WPE-based web browser source element
Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved dmabuf import/export
Hardware-accelerated Nvidia video decoder gained support for VP8/VP9 decoding, whilst the encoder gained support for H.265/HEVC encoding.
Many improvements to the Intel Media SDK based hardware-accelerated video decoder and encoder plugin (msdk): dmabuf import/export for zero-copy integration with other components; VP9 decoding; 10-bit HEVC encoding; video post-processing (vpp) support including deinterlacing; and the video decoder now handles dynamic resolution changes.
ASS/SSA subtitle overlay renderer can now handle multiple subtitles that overlap in time and will show them on screen simultaneously
Meson build feature-complete (with the exception of plugin docs) and it is now the recommended build system on all platforms. The Autotools build is scheduled to be removed in the next cycle.
GStreamer Rust bindings and Rust plugins module
GStreamer Editing Services allows directly playing back serialized edit list with playbin or (uri)decodebin
Setting up a ec2 instance on AWS for web real time communication platform over nodejs and socket.io using WebRTC .
Primarily a Web Call , Chat and conference platform uses WebRTC for the media stream and socketio for the signalling . Additionally used technologies are nosql for session information storage , REST Apis foe getting sessions details to third parties.
Below is a comprehensive setup if ec2 t2.micro free tier instance , installation with a webrtc project module and samples of customization and usuage .
Technologies used are listed below :
Server
ec2 instance t2.micro covered under free tier
domain name
SSL certificate
Core module for Web Calling feature
WebRTC
Node.js
socket.io
UI components
javascript
css
html5
bootstrap
jquerry
Supporting setup for session management
Code version-ing and maintenance
git
npm
Amazon’s free tier ec2
Amazon EC2
ec2 instances are elastic compute general purpose storage servers that mean that they can resize the compute capacity in the cloud based on load .
750 hours per month of Linux, RHEL, or SLES t2.micro instance usage
Expires 12 months after sign-up.
Some other products are also covered under free tier which may come in handy for setting up the complete complatorm .Here is a quick summary
1.Amazon S3
it is a storage server. Can be used to store media file like image s, music , videos , recorded video etc .
2.Amazon RDS
It a relational database server . If one is using mysql or postgress for storing session information or user profile data . It is good option .
3.Amazon SES
email service. Can be used to send invites and notifications to users over mail for scheduled sessions or missed calls .
4.Amazon CloudFront
It is a CDN ( content delivery network ) . If one wants their libraries to be widly available without any overheads . CDN is a good choice .
Server Setup
Set up environment by installing nvm , npm and git ( source version control)
Since 2015 it has become mandatory to have only https origin request WebRTC’s getUserMedia API ie Voice, video, geolocation , screen sharing require https origins.
Note that this does not apply to case where its required to only serve peer’s media Stream or using Datachannels . Voice, video, geolocation , screen sharing now require https origins
For A POC purpose here is th way of generating a self signed certificate
Transport Layer Security and/or Secure Socket Layer( TLS/SSL) is a public/private key infrastructure.Following are the steps
1.create a private key
openssl genrsa -out webrtc-key.pem 2048
create https certificate using self generate or purchased SSL certificates using fs , node-static and https modules . To know how to create self generated SSL certificates follow section above on SSL certificates.
var fs = require(‘fs’);
var _static = require(‘node-static’);
var https = require(‘https’);
var file = new _static.Server(&amp;amp;amp;amp;amp;amp;quot;./&amp;amp;amp;amp;amp;amp;quot;, {
cache: 3600,
gzip: true,
indexFile: &amp;amp;amp;amp;amp;amp;quot;index.html&amp;amp;amp;amp;amp;amp;quot;
});
Web servers work with the HTTP (and HTTPS) protocol which is TCP based. As a genral rule TCP establishes connection whereas UDP send data packets
Scoketio signalling server as npm
Socket.io determines which of the following real-time communication method is suited to the particular client and its network bandwidth .
WebSocket
Adobe Flash Socket
AJAX long polling
AJAX multipart streaming
Forever Iframe
JSONP Polling
The socket.io server needs a HTTP Server for initial handshake.
The general steps for socketio signalling server are:
1.require socket.io and keep the reference. like
var io = require(‘socket.io’)
2.Create your http / https server
outline in section on webserver
3.bind your http and https servers (.listen)
io.listen(app, {
log: false,
origins: ‘*:*’
});
4. Optionally set transport
io.set(‘transports’, [
‘websocket’
]);
4.setup io events as
io.sockets.on(‘connection’, function (socket) {
//Do domething
});
Note that Socket.io or websockets require an http server for the initial handshake.
&amp;lt;pre&amp;gt;Install ssocketio npm module&amp;lt;/pre&amp;gt;&amp;lt;pre&amp;gt;
npm install socket.io
[/sourcecode ]
Complete code for signalling server
var io = require(‘socket.io’).listen(app, {
log: false,
origins: ‘*:*’
});
io.set(‘transports’, [
‘websocket’
]);
var channels = {};
io.sockets.on(‘connection’, function (socket) {
console.log(&amp;amp;amp;amp;amp;amp;quot;connection &amp;amp;amp;amp;amp;amp;quot;);
var initiatorChannel = ”;
1.Opening page https://<web server ip>:< web server port>/index.html says insecure
This is beacuse the self signed certificates produced by open source openSSL is not recognized by a trusted third party Certificate Agency.
A CA ( Certificate Authority ) issues digital certificate to certify the ownership of a public key for a domain.
To solve the access issue goto https://<web server ip>:< web server port> and given access permission such as outlined in snapshot below
2.Already have given permission to Web Server , page loads but yet no activity .
if you open developer console ( ctrl+shift+I on google chrome ) you will notice that there migh be access related errros in red .
If you are using different server for web server and signalling server or even if same server but different ports you need to explicity go to the signalling server url and port and give access permission for the same reason as mentione above.
3.no webcam capture on opening the page
This could happen due to many reasons
page is not loaded on https
browser is not webrtc compatible
Media permission to webcam are blocked
the machine does have any media capture devices attached
Driver issues in the client machine while accessing webcams and mics .
This article show the different ways to make calls to Wowza Media Engine from external applications and environments for various purposes such as getting server status , listeners , connections , applications and its streams etc .
HTTP Providers
HTTP Providers are Java classes that are configured on a per-virtual host basis.
Some pre packaged HTTP providers that return data in XML :
1. HTTPConnectionCountsXML
Returns connection information like Vhost , application , application instance , message in bytes rate , message out byte rates etc.
Each HTTP provider can be configured with different request filter and authentication method ( none , basic , digest). We can even create our own substitutes for the HTTP providers as defined in the next section .
extending HTTProvider2Base
The following code snippet describes the process of creating a Wowza Web services that return a json containing all the values .
Imports to build a HTTP provider
import com.wowza.wms.application.*;
import com.wowza.wms.vhost.*;
import com.wowza.wms.http.*;
import com.wowza.wms.httpstreamer.model.*;
//since we want to return in json format
import org.json.simple.JSONObject;
The class declaration is as folllows
public class DCWS extends HTTProvider2Base
{
....
}
The code to extract application names
public JSONObject listChannels(){
JSONObject obj=new JSONObject();
//get params from virtual host and iterate through it
List&lt;String&gt; vhostNames = VHostSingleton.getVHostNames();
Iterator&lt;String&gt; iter = vhostNames.iterator();
while (iter.hasNext())
{
String vhostName = iter.next();
IVHost vhost = (IVHost)VHostSingleton.getInstance(vhostName);
List&lt;String&gt; appNames = vhost.getApplicationNames();
Iterator&lt;String&gt; appNameIterator = appNames.iterator();
int i=0;
while (appNameIterator.hasNext())
{
String applicationName = appNameIterator.next();
try {
String key = "channel"+ (++i);
obj.put(key, URLEncoder.encode(applicationName, "UTF-8"));
}
catch (UnsupportedEncodingException e) {
e.printStackTrace();
}
}
}
return obj;
}
XMPP is a open XML technology for real-time communication. Applications are instant messaging, presence, media negotiation, whiteboarding, collaboration, lightweight middleware, content syndication, and generalized XML routing according to XMPP standards Foundation (XSF) .
Extensible Messaging and Presence Protocol (XMPP) is a communications protocol for message-oriented middleware based on XML (Extensible Markup Language). – wikipedia
XMPP Server
Some popular servers on XMPP are ejabbred ( written in erlang licensed by GPL2) and openfire ( written in Java licensed by Apache ). This article will show the installation steps for openfire on Ubuntu version 15 64 bit system
3. Goto bin and run openfire server with ./openfire start
4. Gotot the web admin url http://localhost:9090/ . For first time the setup screen will appear
5. Proceed with installation .
It will show screens to select the mysql driver and database . Create a empty db name called openfiredb and add that to mysql url in setup screen of openfire
It will also request a administrator username and password I choose to give admin admin as the username and password alike .
6. change the interface inside of openfire.xml file in location /opt/openfire/conf
7. After the installation login to the server admin console with the admin username and password which is admin admin in our case
8. Review the server settings etc from the admin web console
9. Incase the server setup did not go as planned we can reinstall the server again by dropping the database , creating a fresh empty database and modifying the following from true to false in openfire.xml file in location /opt/openfire/conf
<setup>true</setup>
Test the XMPP Server Installation using Spark client
1.Spark can also be downloaded from the same url as was used to download server . Choose your operating system for download
2.Register a spark client with the server
3. after registering the client presence should be indicated in the user summary by online status
4.Register another client with the same conf except username and password and perform messaging between them
XMPP Java Client
Source Code for a Simple Java Application using Smack4 communicating with XMPP servers
package testxmppsmack;
import java.io.IOException;
import org.jivesoftware.smack.ConnectionConfiguration.SecurityMode;
import org.jivesoftware.smack.SmackException;
import org.jivesoftware.smack.XMPPException;
import org.jivesoftware.smack.SmackException.NotConnectedException;
import org.jivesoftware.smack.chat.Chat;
import org.jivesoftware.smack.chat.ChatManager;
import org.jivesoftware.smack.chat.ChatMessageListener;
import org.jivesoftware.smack.packet.Message;
import org.jivesoftware.smack.tcp.XMPPTCPConnection;
import org.jivesoftware.smack.tcp.XMPPTCPConnectionConfiguration;
public class JabberSmackAPI {
public static void main(String argsp[]){
XMPPTCPConnectionConfiguration config = XMPPTCPConnectionConfiguration.builder()
.setServiceName("machine")
.setUsernameAndPassword("admin", "admin")
.setCompressionEnabled(false)
.setHost("127.0.0.1")
.setPort(5222)
.setSecurityMode(SecurityMode.disabled)
/* .setSecurityMode(SecurityMode.required) keep this commented */
.setSendPresence(true)
.build();
// Create a connection to the the local XMPP server as defined in config above.
XMPPTCPConnection con = new XMPPTCPConnection(config);
// Connect to the server code is encapsulated in try/catch block for exception handling
try {
con.connect();
System.out.println("Connected "+con.isConnected());
} catch (SmackException | IOException | XMPPException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
}
//Login before performing other tasks like messaging etc
try {
con.login("altanai", "aaa");
System.out.println("Loggedin "+con.isAuthenticated());
} catch (XMPPException | SmackException | IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
// Start a new conversation with another account holder caled altanaibisht ( I created 2 user accounts one with my first name and another with fullname)
Chat chat = ChatManager.getInstanceFor(con).createChat("altanaibisht@localhost");
try {
chat.sendMessage("Did you try out the new code i send you last night ?");
System.out.println("Chat Send ");
} catch (NotConnectedException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
// Disconnect from the server
con.disconnect();
}
}
Some errors and their resolution while building and running the above code as Java Application are as follows :
1. Cannot instantiate XMPPConnection
Use XMPPTCPConnection instead of XMPPConnection in Smack 4.
9. org.jivesoftware.smack.SmackException: javax.net.ssl.SSLHandshakeException: sun.security.validator.ValidatorException: PKIX path building failed: sun.security.provider.certpath.SunCertPathBuilderException: unable to find valid certification path to requested target
.setSecurityMode(SecurityMode.disabled)
Once the program build and runs succesfully connecting to the XMPP server ( which is running ofcourse ) , open a sapark client and test the application with it.
Summary
An alternative to XMPP messaging is the SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) based on Session Initiation Protocol (SIP).
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