H323

H.323 is an ITU-T recommendation that specifies how multimedia traffic is carried over packet networks. H.323 utilizes existing standards (Q.931, for example) to accomplish its goals. H.323 is a rather complex protocol that was not created for simple development of applications. Rather, it was created to enable multimedia applications to run over "unreliable" data networks. Voice traffic is only one of the applications for H.323. Most of the initial work in this area focused on multimedia applications, with video and data-sharing a major part of the protocol.

Applications require significant work if they are to be scalable with H.323; for example, to accomplish a call transfer requires a separate specification (H.450.2). SGCP and MGCP, on the other hand, can accomplish a call transfer with a simple command, known as a modify connection (MDCX), to the gateway or endpoint. This simple example represents the different approaches built into the protocol design itself—one tailored to large deployment for simple applications (MGCP), and the other tailored to more complicated applications but showing limitations in its scalability (H.323).

To further demonstrate the complexity of H.323, Figure 1-13 shows a call-flow between two H.323 endpoints.

Figure 1-13. H.323 Call-Flow

Figure 1-13. H.323 Call-Flow

Figure 1-13 illustrates the most basic H.323 call-flow. In most cases, more steps are needed because gatekeepers are involved.

To better explain Figure 1-13, let's step through the call-flow:

1. Endpoint A sends a setup message to Endpoint B on TCP Port 1720.

2. Endpoint B replies to the setup message with an alerting message and a port number to start H.245 negotiation.

3. H.245 negotiation includes codec types (G.729 and G.723.1), port numbers for the RTP streams, and notification of other capabilities the endpoints have.

4. Logical channels for the UDP stream are then negotiated, opened, and acknowledged.

5. Voice is then carried over RTP streams.

6. Real Time Transport Control Protocol is used to transmit information about the RTP stream to both endpoints.

This call-flow is based on H.323 v1. H.323 v2, however, enables H.245 negotiation to be tunneled in the H.225 setup message. This is known as fast-start , and it cuts down on the number of roundtrips required to set up an H.323 call. It does not, however, make the protocol any less complex. More detailed analysis of H.323 is found in Chapter 10.

SGCP and MGCP

SGCP and MGCP were developed to enable a central device, known as a Media Gateway Controller (MGC) or soft-switch, to control endpoints or Media Gateways (MGs). Both of those protocols are referenced simultaneously as xGCP . You can develop applications through the use of standard-based APIs that interface with the MGCs and offer additional functionality (such as call waiting and CLASS features) and applications.

The Cisco version of this technology is known as the Virtual Switch Controller (VSC). In this scenario, the entire IP network acts like one large virtual switch, with the VSC controlling all the MGs.

Figure 1-14 shows how a typical network design works with a virtual switch running MGCP.

Figure 1-14. Virtual Switch Controller

Figure 1-14. Virtual Switch Controller

Figure 1-14 also shows how the legacy PSTN and enterprise networks are connected to gateways or endpoints that enable access into the new packet network. This packet gateway receives direction from the Call Agent (VSC), which can communicate with the SS7 network and the IN and can tell the gateways or endpoints how and when to set up the call.

Figure 1-14 also shows how the legacy PSTN and enterprise networks are connected to gateways or endpoints that enable access into the new packet network. This packet gateway receives direction from the Call Agent (VSC), which can communicate with the SS7 network and the IN and can tell the gateways or endpoints how and when to set up the call.

To understand Figure 1-14 in greater detail, all the various components must be described. The existing PSTN/SS7 network is connected to the Switching Transfer Point (STP), which also is connected to the MGC or Call Agent. This connection is where the signaling (SS7) takes place.

The PSTN/SS7 network is also connected to an MG, which is a signal-less trunk that is often known as an Inter-Machine Trunk or IMT. The MG is where the 64-kbps voice trunks are converted into packets and placed onto the IP network.

The MGCs or Call Agents also intercommunicate. This protocol is currently undefined in the standards bodies. Based on the current state of the industry, however, it appears that a variant of SIP or ISDN User Part (ISUP) over IP—a portion of SS7 running on top of IP—will be the primary protocol. The MGCs have a connection to the IN (described earlier in this chapter) to provide CLASS services. The MGCs receive signals from the SS7 network and tell the MGs when to set up IP connections and with which other MGs they should set them up.

The MG on the right side of Figure 1-14 does not have a connection to the SS7 network. Therefore, a mechanism known as signaling backhaul must be used to tell the VSC when and how a call is arriving. Signaling backhaul is normally done with ISDN. The MG or some other device separates the D channel from the B channels and forwards the D channel to the MGC through IP. Signaling backhaul is currently undefined in the standards bodies. By the time this book is printed, however, there should be a specification for signaling backhaul.

For a more detailed explanation of how all these components work together, see Chapter 12, "Gateway Control Protocols" and Chapter 13, "Virtual Switch Controller."

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