- Hosted PBX
- Event services for Line status notifications
- Routing services
- Call Control Manager (CCM)
- Call processing framework (CPF)
- Multiple shared Lines and their statuses
- Clustering Call control managers
- Call Replication and distributed memory management
- Controlling Line Calls using AOR and Resource Zones
- Incoming request Zone vs Active Resource Zone
- Bridges for Local Media connections
- Interzone proxy Handler
- Dial Trunk using multiple dedicated SIP lines and connect via Media Bridge
- Call Diversion
- Call barge
- Hold-Resume and Music on Hold in multi-line evironment
- Session Border Controllers (SBC) for PBX
- SIP to PSTN
- SIP trunking
- support for various fixed or mobile endpoints
- NAT (Network address translator)
- Load balancing
- Security, QoS and Regulatory compliance
- Terminating at carriers, PSTN and IP gateways
- PBX connection to IMS systems
- VoIP with PBX without IMS
- VoIP with PBX connected to IMS
- Additional SBC features
- UseCase Scenarios
- Future of IP PBX
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.
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 calls within enterprise and hardphone sites read more about resource zones later in the article
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
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.
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.
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.
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.
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. They can be used for various roles such as
- 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 .
There are 2 ways to integrate IP calls to telecom provider endpoints such as GSM or LTE phones.
- PRI lines
- SIP trunks
Additional SBC features
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.
Typical VOIP/SIP scenario without IMS
A basic enterprise VoIP/SIP solution is illustrated in Figure. The key element is a soft switch (SIP PBX) which might be implemented as a combination of several SIP entities, such as SIP registrar, proxy server, redirect server, forking server, Back-To-Back User Agent (B2BUA) etc. SIP clients can be SIP hard-phones or soft-phones on PCs, PDAs etc. A PSTN gateway links the enterprise SIP PBX to the public PSTN. Enterprise applications, media servers, presence servers, and the VoIP/SIP PBX are interconnected through a company intranet.
VoIP System with IMS : With IMS, applications will be able to establish sessions across different access networks, with guaranteed QoS, flexible charging & AAA support. Call control, user’s database and services, which are the typical functions of softswitch, are controlled by separate units in IMS. CSCF (Call Session Control Function) handles session establishment, modification and release of IP multimedia sessions using the SIP/SDP protocol suite. Services features are separated from call control and handled by application servers. Subscriber’s database function is separated from service logic function and handled by HSS using open subscriber directory interface.
Link registration using subscribe-notify can be handled via Enterprise App server in PBX.
Forking proxy Setup of PBX : The enterprises SIP PBX can work as a forking proxy during call setup to redirect the calls.
Other usecases can involve presence sharing between different enterprise PBX with both domains interconnect their presence servers.
Any VOIP dependant system which deals with bulksome voice / video traffic from external endpoints is a usages scenarios. Listing few
- provision of pre-defined enterprise based SIP URI.
- 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
The There are many more features and usecases for an IP-PBX solution for an enterprise. The features of modern IP PBX systems are a big addon to internal secure telecom channel in an company and accross its various office.
There has been a significant shift in replacing hard PBX systems with software-based IP PBX such as using Freeswitch, Asterisk or other commercial-grade SIP servers which seamlessly integrate into other business software such as CRM systems, task force management systems.
In recent times cloud telephony providers, particularly CPaaS platforms have revolutionized the IP telecommunication landscape with lightweight and feature-rich communication agents( web, native platform) and services such as programmable API to control call logic and services such as recording, IVR announcements, call parking, Automatic Queueing so on.