SS7 and SIP/H.323 Interworking

The ITU-T originally developed the H.323 [125] for multimedia over Local Area Networks (LANs). It is not a single protocol; rather, it is a vertically-integrated suite of protocols that define the components and signaling. Though it was originally used for video-conferencing, H.323 was enhanced to better support VoIP with the Version 2 release. It is currently the most widely-deployed VoIP solution today.

One of the main complaints about H.323 is its complexity. With H.323, many messages must be passed to set up even a basic voice call. SIP [124], is considered a simpler, more flexible alternative to H.323. SIP is a signaling protocol that handles the setup, modification and teardown of multimedia sessions. It was developed in the IETF as a signaling protocol for establishing sessions in an IP network. A session can be a simple two-way telephone call or a multimedia conference. SIP is becoming a popular favorite as the future of VoIP.

So, how does SigTran play a role in H.323 and SIP? SigTran can provide PSTN connectivity to H.323 and SIP networks. A PSTN Gateway application can be used to fulfill this need. The PSTN Gateway sits on the edge of the circuit-switched and packet-switched networks and provides SIP or H.323 interworking to SS7 in the PSTN. Figure 14-29 shows an example of an SIP PSTN Gateway application. In this example, the MGC connects to the SGs using SigTran.

Figure 14-29. SIP-PSTN Gateway Application

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Figure 14-30 shows a similar example of an H.323 PSTN Gateway application.

Figure 14-30. H.323-PSTN Gateway Application

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Another interesting application is the PSTN transit application, in which calls originate and terminate on TDM interfaces and then transit a voice packet network (such as SIP or H.323). Service providers can use this application to offload their tandem and transit Class 4 and Class 3 switches. This application creates the need for an ISUP transparency. SIP-T [150] (SIP for Telephones) provides a framework for the integration of the PSTN with SIP. Figure 14-31 shows an example of using SIP-T for a PSTN transit application.

Figure 14-31. SIP Transit Application

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SIP-T meets the SS7 to SIP interworking requirements by providing the following functions:

• A standard way of mapping ISUP information into the SIP header for calls that originate in the PSTN. This function ensures that the SIP contains sufficient information to route calls (for example, in the case where routing depends on some ISUP information).

• Use of the SIP INFO [151] Method to transfer mid-call ISUP signaling messages.

• A means for MIME [152] encapsulation of the ISUP signaling information in the SIP body provides for ISUP transparency.

When the MGC receives an ISUP message, the appropriate ISUP parameters are translated to the SIP header fields and the ISUP message is encapsulated in a MIME attachment, which intermediate SIP entities treat as an opaque object. If the SIP message terminates the call, it ignores the ISUP attachment because it has no need for it. However, if the call terminates on the PSTN, the encapsulated ISUP message is examined and used to generate the outgoing ISUP message. The version parameter included in the MIME media type information indicates the encapsulated ISUP message's ISUP variant. If there are different ISUP variants on the origination and termination side, it is up to the terminating MGC to perform ISUP translation between the variants.