An Overview of SIP

 

SIP on an IMG

Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call whereby an IMG can be used as a Media Gateway to allow two separate networks to connect. The IMG supports SS7 to SIP, ISDN to SIP, CAS to SIP, SIP to SIP, and H.323 to SIP. Below is exemplary diagram of an IMG in a TDM to IP network.

 

dg_img_sip_network_dialogi_logo.png

 

Supported SIP Features

 

RFC

Description

2246

Transport Layer Security (TLS) for SIP

2327

Session Description Protocol (SDP)

2976

SIP Info

3240

Internet media type message/sipfrag

3261

SIP: Session Initiation Protocol

3262

SIP PRACK

3263

Locating SIP servers for DNS lookup SRV and A records

3264

SDP Offer/Answer Model (Do not support multiple 'm' lines in SIP SDP)

3265

SIP Subscribe/Notify

3311

SIP Update

3323

SIP Privacy Header

3325

Asserted Identity

3326

SIP Reason Header

3332

M3UA Adaption Layer

3372

SIP for Telephones (SIP-T/SIP-I)

3398

ISUP/SIP Mapping

3515

SIP Refer

3551

Payload Type Support

3578

ISUP Overlap Signaling to SIP

3581

Symmetric Response Routing

3666

Call Flows - SIP to PSTN Dialing

3711

IP Media Layer Security Standard (RTP/RTCP)

3725

Third Party Call Control for SIP

3764

ENUM for SIP Address of Record

3891

SIP Replace Header

3892

SIP Referred by Mechanism

4028

SIP Session Timer

4040

Clear Channel Codec Support

4244

SIP History info (for call diversion)

4568

IP Signaling Layer Security Standard (RTP/RTCP)

4904

Trunk Group Parameter Support

 

 

Basic Support

 

Supported Methods

 

SIP Extensions

 

Routing/Call Handling

 

Media

Note: If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message, the gateway returns the same rate in the response SDP.

 

Interworking

Basic Support:

 

 

SIP