At the time of writing this article on SIP and related VOIP technologies I a newbie in VOIP domain , probably just out college . However over the past decade , looking at the steady traffic to these articles , I have tried updating the same with new RFC standards and market trends .
In this updated version (2019) , the main points described are
- SIP transactions , dialog , branch
- Record Routing
- strict routing
- loose routing
- System Components in SIP based Voip ( Requests and Responses )
- SIP Transport Layer
- Session Description Protocol (SDP)
- Mobility and Location Service
- Network Address Translator ( NAT)
- SIP Call Flows
- Call Redirection
- click to Dial
- SIP for Instant Messaging and Presence Leveraging Extensions ( SIMPLE)
The Session Initiation Protocol (SIP) is a multimedia signalling protocol that has evolved the defacto communication standard for IP telephony.
Even today it forms the primary protocol for many Real Time Communication platforms which are integrated with telecom carriers and provide Cloud and IP based Services for applications such as robo/mass calls for advertising, API based calls like OTP generator, IVR announcements with DTMF input like customer care centre etc. Infact it would be not far from truth to say that converged platform we find today are a result of SIP integrating with the IP world.
Converged platforms integrates audio, video, data, presence, instant messaging, voicemails and conference services into a single network . SIP is the key component to build an advanced converged IP communication platform or rich multimedia Real time communication service.
SIP can be used to create programmable APIs and complex call routing VoIP scripts such as PBX , SBC etc.
Bears the support of many high quality open source and freeware SIP client , servers , proxies , tool such as Kamailio , Astersk , Freeswitch , Sipp , JAINSIP etc .Also supported on most standardised VoIP hardware and network such as Cisco, Microsoft, Avaya, and Radvision.
It is standardized by Internet Engineering Task Force (IETF) such as RFC 3261 which describes SIP v2 . Architecturally SIP request response ( 404 , 301 ) format is very similar to HTTP and its addressing schemes have a resemblance to SMTP ( sip:email@example.com) .
SIP ( Session Initiation Protocol) negotiates session between 2 parties. It primarily exchanges headers that are used for making a call session such as example of outgoing telephone call from SIP session invite .
Session Initiation Protocol (INVITE) Request-Line: INVITE sip:firstname.lastname@example.org;transport=tcp SIP/2.0 Method: INVITE Request-URI: email@example.com;transport=tcp Request-URI User Part: altanai Request-URI Host Part: telecomcompany.com [Resent Packet: False] Message Header Via: SIP/2.0/TCP 126.96.36.199:5080;rport;branch=z9hG4bKceX7a2H2866cN Transport: TCP Sent-by Address: 188.8.131.52 Sent-by port: 5080 RPort: rport Branch: z9hG4bKceX7a2H2866cN Max-Forwards: 41 From: "+16014801797" <sip:+firstname.lastname@example.org>;tag=7HKgjNQ6y2FSj SIP Display info: "+16014801797" SIP from address: sip:+email@example.com SIP from address User Part: +16014801797 E.164 number (MSISDN): 16014801797 Country Code: Americas (1) SIP from address Host Part: 184.108.40.206 SIP from tag: 7HKgjNQ6y2FSj To: <sip:firstname.lastname@example.org;transport=tcp> SIP to address: sip:email@example.com;transport=tcp SIP to address User Part: altanai SIP to address Host Part: telecomcompany.com SIP To URI parameter: transport=tcp Call-ID: e10306be-0cfd-4b38-af3c-b2ada0827cef CSeq: 126144925 INVITE Contact: <sip:firstname.lastname@example.org:5080;transport=tcp> User-Agent: phone1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 249 SIP Display info: "+16014801797" SIP PAI Address: sip:+email@example.com
The SIP philosophy :
- reuse Internet addressing (URLs, DNS, proxies)
- utilize rich Internet feature set
- reuse HTTP coding
- text based
- makes no assumptions about underlying protocol:
TCP, UDP, X.25, frame, ATM, etc
- support of multicast
SIP URI can either be in format of sip:firstname.lastname@example.org (RFC 2543 ) or sips:email@example.com ( secure with TLS over TCP RFX 3261) . Additionally SIP URI resolution can either be
- DNS SRV based such as firstname.lastname@example.org with SIP servers locating record for domain “telecomcompnay.com ” or
- FQDN ( Fully qualified domain name ) / contact / ip address based such as email@example.com or altanai@us-west1-prod-server . Both of which do not need any resolution for routing.
Tags are pseudo-random numbers inserted in To or From headers to uniquely identify a call leg
Max forwards is a count decremented by each proxy
that forwards the request.When count goes to zero, request is discarded and 483 Too Many Hops response is sent.Used for stateless loop detection.
Content-Type indicates the type of message body attachment. In this case application /SDP but others could be text/plain, application/cpl+xml, etc.)
Content-Length indicates the octet (byte) count of the message body
Contact direct route to contact the sender, composed of SIPURI with a user name and IP or FQDN. USed for later requests to directly reach the destination such as ACK after INVITE
via gives the last SIP hop as IP, transport, and transaction-specific parameters along with branch that identifies the transaction
each proxy adds an additional via header. fianlly via header is used to route back the responses . This ensures the user agents after the initial request dont have to rely on DNS and location tables to route the messages.
Firewalls can sometimes block SIP packets , change TCP to UDP or change IP address of the packets. Record-Route can be used , ensures Firewall proxy stays in path . Clients and Servers copy Record-Route and put in Route header for all messages
Message body is separated from SIP header fields by a blank line (CRLF).
A SIP transaction occurs between a UAC and a UAS in form of 1 request , its provisional and final response.
All transactions are independent of each other. Each transaction are uniquely identified by the branch id on the via header and the cseq.
Via: SIP/2.0/UDP <server ip>:5060;branch=z9hG4bKcb16.c47db56d6d8eb62677a0f0dc733cd73d.0 ... CSeq: 1 INVITE
Each transaction is uniquely identified by: the branch-id on the Via-header and the Cseq header
for ACK given below , tid=-d8754z-deea18278a05ce16-1—d8754z-
T 2017/06/06 06:56:03.656614 :37126 -> :5060 [AP] ACK sip:9876543210@:5080;transport=tcp SIP/2.0. Via: SIP/2.0/TCP :38834;branch=z9hG4bK-d8754z-deea18278a05ce16-1---d8754z-;rport. Max-Forwards: 70. To: :5080>;tag=fdc0b562c1d44395f53d16b622397a3f-589d. From: >;tag=b5327b03. Call-ID: MTllYjkyZjczMjhjM2I5OGE4MTgzZDUxODVjYmM0YzY. CSeq: 1 ACK. Content-Length: 0.
For CANCEL given below , tid=-d8754z-04665556a3f8c928-1—d8754z-
T 2017/06/06 06:53:09.643301 :37126 -> :5060 [AP] CANCEL sip:9876543210@:5080;transport=tcp SIP/2.0. Via: SIP/2.0/TCP :38834;branch=z9hG4bK-d8754z-04665556a3f8c928-1---d8754z-;rport. Max-Forwards: 70. To: :5080>. From: >;tag=c0869612. Call-ID: NTJhMGU1ZTA1NTAyZTYzZmUzMWQ0NjQ2MjIwYTE0MmI. CSeq: 1 CANCEL. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 0.
ACK – For positive replies (2XX), a new transaction is created with new CONTACT header and it can be sent straight to the UAS bypassing the proxy. For negative replies, it stays part of INVITE transaction hence request is sent to the same proxy as INVITE.
The branch parameter is a transaction identifier. Responses relating a request can be correlated because they will contain the same transaction identifier.
The p2p relationship between 2 sip endpoints , containing sequence of transactions.
The initiator of the session that generates the establishing INVITE generates the unique Call-ID and From tag. In the response to the INVITE, the user agent answering the request will generate the To tag. The combination of the local tag (contained in the From header field), remote tag (contained in the To header field), and the Call-ID uniquely identifies the established session, known as a dialog. This dialog identifier is used by both parties to identify this call because there could be multiple calls set up between them.
A dialog is uniquely identified by: Call-ID header , remote-tag and local-tag. Dialog id is different for both ends since local and remote for both ends are different.
Example : Notice the to and from tag ids in INVITE and its 200 ok. The dialog id for invite is , 97576NjQ5MTBlNjVjNDQ0MzFmOTEyZGEzYWJjZjQxYjcyYzc70edc66c. First invite doesnt bear the To tag.
INVITE sip:1234567890@ SIP/2.0 Via: SIP/2.0/UDP :59583;branch=z9hG4bK-524287-1---22728813bce01a15;rport Max-Forwards: 70 Contact: :59583> To: > From: >;tag=70edc66c Call-ID: 97576NjQ5MTBlNjVjNDQ0MzFmOTEyZGEzYWJjZjQxYjcyYzc CSeq: 1 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 5.5.0 stamp 97576 Content-Length: 210 v=0 o=- 1559804173873191 1 IN IP4 s=X-Lite release 5.5.0 stamp 97576 c=IN IP4 t=0 0 m=audio 49750 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv
The dialog id, with reversed to and from tag is 97576NjQ5MTBlNjVjNDQ0MzFmOTEyZGEzYWJjZjQxYjcyYzcStNBKgjjXS84r70edc66c
SIP/2.0 200 OK Via: SIP/2.0/UDP :59583;branch=z9hG4bK-524287-1---22728813bce01a15;rport=10973;received= From: >;tag=70edc66c To: >;tag=StNBKgjjXS84r Call-ID: 97576NjQ5MTBlNjVjNDQ0MzFmOTEyZGEzYWJjZjQxYjcyYzc CSeq: 1 INVITE Contact: :5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.9.0-742-8f1b7e0~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 222 Remote-Party-ID: "1234567890" >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1559778909 1559778910 IN IP4 s=FreeSWITCH c=IN IP4 t=0 0 m=audio 25266 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
All requests sent within a dialog are by default sent directly from one user agent to the other. Only requests outside a dialog traverse SIP proxies. This approach makes SIP network more scalable because only a small number of SIP messages hit the proxies.
However few request need to explicitly state that they need to stay on path of proxies such as for accounting during termination of when NAT process is being carried out then . For these we need to insert a Record-Route header field into SIP messages which contain address of the proxy. Messages sent within a dialog will then traverse all SIP proxies that put a Record-Route header field into the message.
The server copies the Record-Route header field unchanged into the
response. (Record-Route is only relevant for 2xx responses. ) ie the end point recipient will also mirror the proxies for the response.
Rewrite the Request-URI ie Request-URI always contained URI of the next hop so it is necessary to save the original Request-URI as the last Route header field. Defined in RFC2543
Request-URI is no more overwritten, it always contains URI of the destination user agent, therby keeping target seprated from route. ( ;lr) . If there are any Route header field in a message, then the message is sent to the URI from the topmost Route header field. Defined in RFC 3261
Components of SIP based VoIP Solution
SIP Request methods :
- INVITE : Initiates negotiation to establish a session ( dialog). Usually contains SDP payload. Another invite during an existing session ( dialog) is called an RE-INVITE. A RE-INVITE can be used for
- hold / resume a call
- change session parameters and codecs in mid of a call
- ACK : Acknowledge an INVITE request by completing the 3 way handshake . If an INVITE did not contain media contain then ACK must contain it .
- BYE : Ends a session ( dialog).
- CANCEL : Cancels a session( dialog) before it establishes .
- REGISTER : Registers a user location (host name, IP) on a registrar SIP server.
- OPTIONS : Communicates information about the capabilities of the calling and receiving SIP phones ( methods , extensions , codecs etc )
- PRACK : Provisional Acknowledgement for provisional response as 183 ( session in progress) . PRACK only application to 101- 199 responses .
- SUBSCRIBE : Subscribes for Notification from the notifier. Can use Expire=0 to unsubscribe.
- NOTIFY : Notifies the subscriber of a new event.
- PUBLISH : Publishes an event to the Server.
- INFO : Sends mid session information.
- REFER : Asks the recipient to issue call transfer.
- MESSAGE : Transports Instant Messages.
- UPDATE : Modifies the state of a session ( dialog).
Some SIP responses :
1xx = Informational SIP Responses
183 Session Progress
2xx = Success Responses
200 OK – Shows that the request was successful
3xx = Redirection Responses
4xx = Request Failures
404 Not Found
405 Method Not Allowed
407 Proxy Authentication Required
408 Request Timeout
480 Temporarily Unavailable
481 Call/Transaction Does Not Exist
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
482 Loop Detected
483 Too Many Hops
5xx = Server Errors
500 Server Internal Error
503 Service Unavailable
6xx = Global Failures
600 Busy Everywhere
604 Does Not Exist Anywhere
606 Not Acceptable
SIP callflow diagram for a Call Setup and termination using RTP for media and RTCP for control. Read about SIP messages indepth here
SIP Transport Layers
We know the ISO OSI layers which servers as a standard model for data communications .
- Physical Layer : Ethernet , USB , IEEE 802.11 WiFi, Bluetooth , BLE
- Data Link Layer : ARP ( Address Resolution Protocol ) , PPP ( point to point protocol ) , MAC ( Media Access control ) , ATM , Frame Relay
- Network Layer : IP (IPv4 / IPv6), ICMP, IPsec
- Transport : TCP , UDP , SCTP
- Session : PPTP ( Point to point tunnelling protocol) , NFS, SOCKS
- Presentation : Codecs such as JPEG , GIFF , SSL
- Application : Application level like Call -manager/ softphone as HTTP , FTP , DNS , SIP , RTSP , RTP , DNS
SDP ( Session Description Protocol)
SIP can bear many kinds of MIME attachments , one such is SDP. It uses RTP/AVP Profiles for common media types . Specified by RFC 3264 . It defines media information and capabilities such as codecs , termination points .
Contains connection headers used for establishing the session . Sample SDP payload for Invite SIP above :
Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1532932581 1532932582 IN IP4 220.127.116.11 Owner Username: FreeSWITCH Session ID: 1532932581 Session Version: 1532932582 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 18.104.22.168 Session Name (s): FreeSWITCH Connection Information (c): IN IP4 22.214.171.124 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 126.96.36.199 Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 29398 RTP/AVP 0 101 Media Type: audio Media Port: 29398 Media Protocol: RTP/AVP Media Format: ITU-T G.711 PCMU Media Format: DynamicRTP-Type-101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Sample Rate: 8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Sample Rate: 8000 Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20
v=0 indicates the start of the SDP content.
o=FreeSWITCH 1532932581 1532932582 IN IP4 188.8.131.52 , is session origin and owner’s name
c=IN IP4 184.108.40.206 is connect information Specifies the IP address of a session.
m= is Media type – audio, port – 29398, RTP/AVP Profile – 0 and 101
Attribute profile – 0, codec – PCMU, sampling rate – 8000 Hz and Attribute profile – 101, telephone-event
Authentication , security , confidentiality and integrity form the basic requirement for any communication system . To protect against hacking a user account and Denial of service attacks , SIP uses HTTP digest authentication mechanism with nonces and challenges along with 407 Proxy Authorization required and 401 unauthorised . The sender has to resend the request with MD5 hash of nonce and password ( password id never send in clear ). Thus preventing man-in-middle attacks.
Challenge / Response Scheme :
- Sends REGISTER and receives 401 / 407 Challenge + nonce
- Again sends REGISTER + MD-5 hash (pw + nonce) get a 200 OK
REGISTER using HTTP Digest for authentication using TLS transport, challenge is in form
CSeq: 1 REGISTER WWW-Authenticate: Digest realm="atlanta.example.com", qop="auth", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale=FALSE, algorithm=MD5
challenge response by UA to UAS
Authorization: Digest username="bob", realm="atlanta.example.com" nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sips:ss2.biloxi.example.com", response="dfe56131d1958046689d83306477ecc"
Cancellation of Registration – UA sends REGISTER request with Expires: 0 Contact: * , to apply to all . Since user is already authenticated , it is not challenged again .
To prevent spoofing ie impersonating as server , SIP provides server authentication too. Required by ITSP’s ( Internet telephony service providers ) .
End to end encryption is achieved thorough TS and SRTP. More on SIP Security here .
Mobility and Location Service
According to RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers , if the proxy finds that the request is for an outside domain , it will take help of a DNS server to resolve to IP address of target domain and forward the request. Then target domain proxy used REGISTRAR’s discovery services to find if user is present in the host via location table entry . If found then request reaches the user .
To provide session mobility SIP endpoints send Register request to their respective registrar as they move and update their location. As User changes terminals , they registers themselves to the appropriate server
– Location server tracks the location of user
– Redirect servers prioritise the possible locations of the user
– Users keep same services as located at home server, while mobile
Call is processed by home servers using RECORD-ROUTE
NAT ( Network Address Translator)
Network Address Translator , defined by RFC 3022 to conserve network space as most packets are exchanged inside a private network itself .
All internet users whether they are using Wifi , 3G/LTE, home AP, any other telecom data packet network by TSP or ISP , are assigned a private IP address , which is unreachable from out side world .Addresses are assigned by Internet Assigned Numbers Authority (IANA). Private address blocks are in format of 10.0.0.0/8, 172.16.0.0/12, 192.168.0.0/16.
Therefore when they access the Internet , this address is converted into a globally unique public IP address through a NAT for external communication
SIP Issues around NAT
NATs modify IP addresses (Layer 3)- SIP/SDP are Layer 7 protocols – transparent to NAT
SIP Via:, From: and Contact: headers use not-routable private addresses
SDP states that originator wishes to receive media at not-routable private addresses
If destination on the public internet tries to send SIP or RTP traffic to those private address
Traffic will be dumped by first router
Solution are to use either Application level gateway (ALG) or STUN or Universal Plug and Pray (UPnP)
To rewrite all SIP/SDP source addresses
- SIP Via:, From: and Contact: headers use public NAT address
- SDP addresses use NAT public address
- Use SIP over TCP
Use draft-ietf-sip-symmetric-response-00 and “Symmetric” SIP/RTP
Use same UDP port number for incoming/outgoing
Hold ports open for call duration
Send UDP packet typically every 30 seconds
SIP over UDP uses 30 second re-INVITE, REGISTER or OPTIONs
RTP sends at much higher frequency by default
NAPT ( Network Address Port Translator ) – Can map multiple private IP addresses and ports to one public IP address and ports
To adapt SIP to modern IP networks with inter network traversal ICE, far and near-end NAT traversal solutions are used. Network Address traversal is crtical to traffic flow between private public network and from behind firewalls and policy controlled networks
One can use any of the VOVIDA-based STUN server, mySTUN , TurnServer, reStund , CoTURN , NATH (PJSIP NAT Helper), ReTURN, or ice4j
Near-end NAT traversal
STUN (session traversal utilities for NAT) – UA itself detect presence of a NAT and learn the public IP address and port assigned using NAting. Then it replaces device local private IP address with it in the SIP and SDP headers. Implemented va STUN, TURN, and ICE.
limitations are that STUN doesnt work for symmetric NAT (single connection has a different mapping with a different/randomly generated port) and also with situtatiosn when there are multiple addresses of a end point.
TURN (traversal using relay around NAT) or STUN relay – UA learns the public IP address of the TURN server and asks it to relay incoming packets. Limitatiosn since it handled all incoming and outgong traffic , it must scale to meet traffic requirments and should not become the bottle neck junction or single point of failure.
ICE (interactive connectivity establishment) – UA gathers “candidates of communication” with priorities offered by the remote party. After this client pairs local candidates with received peer candidates and performs offer-answer negotiating by trying connectivity of all pairs, therefore maximising success. The types of candidates
– host candidate who represents clients’ IP addresses,
– server reflexive candidate for the address that has been resolved from STUN
– relayed candidate for the address which has been allocated from a TURN relay by the client.
Far-end NAT traversal
UA is not concerned about NAT at all and communicated using its local IP port. The border controller implies a NAT handling compoenets such as an application layer gateway (ALG) or universal plug and play (UPnP) etc which resolves the private and public network address mapping by act as a back to back user agent (B2BUA).
Far end NAT can also be enabled by deploying a public SIP server which performs media relay (RTP Proxy/Media proxy).
Limitations of this approach
security risks as they are operating in public network
enabling reverse traffic from UAS to UAC behind NAT.
A keep-alive mechanism is used to keep NAT translations of communications between SIP endpoint and its serving SIP servers opened , so that this NAT translation can be reused for routing. It contains client-to-server “ping” keep-alive and corresponding server-to-client “pong” messages. The 2 keep-alive mechanisms: a CRLF keep-alive and a STUN keep-alive message exchange.
Localization Server –Used by the Proxy Server and Redirect Server to obtain the location of the called user (one or more addresses)
Registration Server- Accept registration requests from the client applications . Generally, the service is offered by the Proxy Server or Redirect Server
DNS Server – Used to locate the Proxy Server or Redirect Server using NAPTR or SRV records
The 3 types of SIP URIs,
- address of record (AOR)
- fully qualified domain name (FQDN)
- globally routable user agent (UA) URI
- SIP uniform resource identifiers (URIs) are identified based on DNS resolution since the URI after @ symbol contains hostname , port and protocol for the next hop.
Adding record route headers for locating the correct SIP server for a SIP message can be done by :
DNS service record (DNS SRV)
naming authority pointer (NAPTR) DNS resource record
Steps for SIP endpoints locating SIP server
- From SIP packet get the NAPTR record to get the protocl to be used
- Inspect SRV record to fetch port to use
- Inspect A/AAA record to get IPv4 or IPv6 addresses
ref : RFC 3263 – Locating SIP Servers
Can use BIND9 server for DNS resolution supports NAPTR/SRV, ENUM, DNSSEC, multidomains, and private trees or public trees.
Sending Call invite but as Redirect Server responded with 302 moved temporary , a new destination address is returned. The invite is forwarded to another proxy server which connects the sip endpoints again after consultation with Redirect server .
In this stage of we see the call getting connected to sip endpoint via 2 proxy servers . The redirect server doesnt get into path once the initial sip request is send.
After communication the endpoints send BYE to terminate the session
This callflow deals with the use-case when a user maybe registered from multiple SIP phones ( perhaps one home phone , one car and one office desk etc ) and wants to receive a ring on all registered phone ie fork a call to multiple endpoints .
In the above diagram we can see a forked invite going to both the sip phones . Both of them reply with 100 trying and 180 ringing, but only 1 gets answered by the user .
After one endpoint sends 200 ok and connects with session , the other receiver a cancel from the sip server .
Click to Dial
A web or desktop application which has HTTP can fire a API call which is interpreted by the controller or SIP server and call is fired .
The API can contain params for to and from sip addresses as well as any authentication token that is required for api authentication and validation .
SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE)
- several vendors who intend to implement SIMPLE
- provides for presence and buddy lists
- Instant Messaging in the enterprise
- telephony enabled user lists
Using SIP based Call routing algorithms and flows , one can build carrier grade communication solution . SIP solutions can hook up with existing telecom networks and service providers to be backward compatible . Also has untapped unlimited potential to integrate with any external IP application or service to provide converged , customised control both for signalling and media planes.
- SIP servlets samples : https://github.com/altanai/sip-servlets
- SIP by Henning Schulzrinne Dept. of Computer Science Columbia University New York
- International Institute of Telecommunications 2000-2004
- Introduction to SIP by Patrick Ferriter from ZULTYS
- Internet Draft, IETF, RFC 2543
- NTU – Internet Telephony based on SIP
RFC 3665 – Session Initiation Protocol (SIP) Basic Call Flow Examples
It contains SIP implementation examples such as
SIP Registration – Successful New Registration , Update of Contact List , Request for Current Contact List , Cancellation of Registration , Unsuccessful Registration
SIP Session Establishment – Successful Session Establishment ,Session Establishment Through Two Proxies,Session with Multiple Proxy Authentication ,Successful Session with Proxy Failure, Session Through a SIP ALG,Session via Redirect and Proxy Servers with SDP in ACK , Session with re-INVITE (IP Address Change) , Unsuccessful No Answer ,Unsuccessful Busy, Unsuccessful No Response from User Agent , Unsuccessful Temporarily Unavailable,
RFC 5359 – Session Initiation Protocol Service Examples
It contains description for services like Call Hold , Consultation Hold , Music on Hold ,
Transfer – Unattended , Transfer – Attended , Transfer – Instant Messaging ,
Call Forwarding Unconditional , Call Forwarding – Busy , Call Forwarding – No Answer ,
3-Way Conference – Third Party Is Added , 3-Way Conference – Third Party Joins ,
Call Management (Incoming Call Screening) , Call Management (Outgoing Call Screening) ,
Call Park , Call Pickup , Automatic Redial ,Click to Dial