Glossary: SIP

Session Initiation Protocol, more commonly known as SIP, is a signaling protocol for packet-based networks and is commonly used, along with H.323 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. TelcoBridges Tmedia media gateways and Tdev development platforms support SIP signaling concurrently with SS7, ISDN and other signaling protocols.

SIP signaling stacks are configured for IP applications and for each Tmedia or Tdev unit requiring SIP signaling.

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:

While TelcoBridges media gateways can perform multiple simultaneous functions such as switching and transcoding as well as deliver value-added services such as IVR or conferencing, they can also be configured to perform a single function. In this case, it is possible to configure TelcoBridges media gateway to act as a SIP gateway, or an SBC.

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:

Specification
TMedia SIP stack support
Toolpack API Support
Media Gateway Application Support

RFC 2327 Session Description Protocol

Yes

Complete

Complete

RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

Yes

Complete

Complete

RFC 2976 SIP INFO Method

Yes

Complete

Partial: For DTMF Tones exchange only

RFC 3204 MIME media types for ISUP and QSIG Objects

Yes

Complete

Complete

RFC 3261 Session Initiate Protocol

Yes

Complete

Complete

RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

Yes

Complete

Complete

RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers

Yes

Complete

Complete

RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)

Yes

Complete

Partial ('Indicating capabilities' not supported)

RFC 3265 Session Initiation Protocol (SIP)-Specific Event Notification

Yes

No

No

RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method

Yes

Partial (Only For Session Timer Refresh)

Partial: For Session Timer Refresh and SDP reception following a SIP INVITE that had no body

RFC 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)

Yes

Partial1

Partial1

RFC 3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks

Yes

Partial (No P-Preferred-Identity)

Partial (No P-Preferred-Identity)

RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP)

Yes

Complete

Complete

RFC 3372 Session Initiation Protocol for Telephones (SIP-T)

Yes

Partial (no ISUP MIME bodies Encryption support)

Partial (no ISUP MIME bodies Encryption support)

RFC 3389 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)

Yes

Complete

Complete

RFC 3398 ISUP-SIP Mapping

Yes

Complete

Complete

RFC 3515 Refer Method

Yes

Complete

Complete

RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control

Yes

Partial (For supported audio codecs, single channel)

Partial (For supported audio codecs, single channel)

RFC 3555 MIME Type Registration of RTP Payload Formats

Yes

Partial (For supported audio codecs)

Partial (For supported audio codecs)

RFC 3578 Overlap

Yes

Partial (Conversion of ISUP Overlap Signalling into SIP en-bloc Signalling)

Partial (Conversion of ISUP Overlap Signalling into SIP en-bloc Signalling)

RFC 3581 An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing

Yes

Complete

Complete

RFC 3665 Session Initiation Protocol (SIP) Basic Call Flow Examples

Yes

Partial*

Partial*

RFC 3666 Public Switched Telephone Network (PSTN) Call Flows

Yes

Complete

Complete

RFC 3764 enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record

Yes

Complete

Complete

RFC 3891 "Replaces" Header

Yes

Complete

Complete

RFC 3892 Referred-By Mechanism

Yes

No

No

RFC 4028 Session Timers in the Session Initiation Protocol (SIP)

Yes

Complete

Complete

RFC 4694 Number Portability Parameters for the "tel" URI

Yes

Yes

Partial (relay of rn and npdi SIP<->SS7)

RFC 4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

Yes

Complete

Complete

RFC 5246 - The Transport Layer Security (TLS) Protocol Version 1.2

Yes

Yes

Yes

RFC 5630 - The Use of the SIPS URI Scheme in the Session Initiation Protocol

Yes

Yes

Yes

RFC 5806 Diversion Indication in SIP

Yes

Yes

Unconditional forward scenario

[1] For more information please contact customer support.

Maximum Capacity

Release
SIP SAP
SIP Transport Server
SIP NAP

TMG800

2.2-2.5

4

10

256

2.6+

16

16

512

TMG3200

2.2-2.5

4

10

256

2.6+

16

16

512

TMG7800

2.2-2.5

64 (4/Tmedia)

160 (10/Tmedia)

256

2.6+

256 (16/Tmedia)

256 (16/Tmedia)

512

References

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