Glossary: SIP
Session Initiation Protocol, more commonly known as SIP, is a signaling protocol for packet-based networks and is commonly used to provide signaling for voice over IP (VoIP) communications.
TelcoBridges and SIP
Toolpack provides support for signaling using the Session Initiation Protocol, more commonly known as SIP, for voice over IP (VoIP) communications. SIP may be used in conjunction with various voice codecs for the media component of a call. Transcoding can be done from different voice codecs with a transcoding device. DTMF RFC2833 to SIP Info or inband DTMF conversion can also be done with the transcoding device.
Based upon your system requirements, you can configure a SIP stack to carry signaling traffic over multiple transport servers, which are IP endpoints comprised of: protocol type (UDP, TCP or TLS), port number and IP interface.
A conceptual illustration is provided below:
TelcoBridges' SIP Implementation
TelcoBridges' SIP implementation works on top of a couple of layers, including SIP and TUCL. In the following figure, grey boxes represent entities that need allocation on the TelcoBridges equipment. The TUCL layer is a transport layer used by SIP on our architecture. TUCL presents some advantages over a simple TCP/IP stack. For instance, it adds tracing facilities to any virtual interface.
Supported SIP RFCs
TelcoBridges supports the following RFCs for SIP:
RFC 2327 Session Description Protocol
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3204 MIME media types for ISUP and QSIG Objects
RFC 3261 Session Initiate Protocol
RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers
RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 3265 Session Initiation Protocol (SIP)-Specific Event Notification
RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method
RFC 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)
RFC 3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks
RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP)
RFC 3372 Session Initiation Protocol for Telephones (SIP-T)
RFC 3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
RFC 3398 ISUP-SIP Mapping
RFC 3515 Refer Method
RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control
RFC 3555 MIME Type Registration of RTP Payload Formats
RFC 3578 Overlap
RFC 3581 An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing
RFC 3665 Session Initiation Protocol (SIP) Basic Call Flow Examples
RFC 3666 Public Switched Telephone Network (PSTN) Call Flows
RFC 3764 enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record
RFC 3891 "Replaces" Header
RFC 3892 Referred-By Mechanism
RFC 4028 Session Timers in the Session Initiation Protocol (SIP)
RFC 4694 Number Portability Parameters for the "tel" URI
RFC 4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 5246 - The Transport Layer Security (TLS) Protocol Version 1.2
RFC 5630 - The Use of the SIPS URI Scheme in the Session Initiation Protocol
RFC 5806 Diversion Indication in SIP
References
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