Assume that User A wants to call User B. User A is located within the IP network, served from a residential gateway and User B is located off-net via the PSTN. When User A picks up the phone, a notify off hook message is sent from the residential gateway to the Call Agent. The Call Agent asks the gateway to create a connection on the endpoint that went off hook by sending a create connection command. The gateway acknowledges to the Call agent the create connection command plus provides a session description. The session description contains information required by a third party, in this case the trunking gateway (G6), to send packets toward the newly created connection. The Session Description Protocol (SDP) is used for this and contains such things as User A's IP address, the UDP port to identify the session, packetization parameters such as compression techniques, and a media type such as RTP audio (voice). The trunking gateway responds to the Call Agent providing its own session description.
The Call Agent uses a modify connection to provide the session description from the trunking gateway to the residential gateway. A two-way full duplex communication can now be set up between the residential gateway (IAD, MTA) and the trunking gateway (G6). When a connection is set up between endpoints, RTP (Real-time Transport Protocol) is used. RTP is an IETF standard that provides end- to-end network transport functions for real time applications such as voice, video, and multimedia. RTP runs on top of UDP because it has multiplexing capabilities, and acknowledgement of packet delivery is not required.
When two endpoints are located on gateways that are managed by the same call agent, the creation is done via the following steps:
- The Call Agent asks the first gateway (MG 1) to create a connection on the first endpoint. The gateway allocates resources to that connection, and respond to the command by providing a session description that contains IP address, UDP port, etc.
- The Call Agent asks the second gateway to create a connection on the second endpoint. The command carries the session description provided by the first gateway. The gateway allocates resources to that connection, and respond to the command by providing its own session description.
- The Call Agent uses a ModifyConnection command to provide this second session description to the first endpoint. Once this is done, communication can proceed in both directions.
The Call Agent removes a connection by sending to the gateway a DeleteConnection command. The gateway may also, under some circumstances, inform a gateway that a connection could not be sustained.
Figure 2. MGCP call setup.
MGCP Applications
MGCP architecture allows for specialization of function and economies of scale and is expected to become the architecture of choice in next generation converged voice/data IP networks.
Currently, MGCP and Session Initiation Protocol (SIP) are the two carrier-class interoperability protocols with the most promise of becoming industry standards. The inherent simplicity of these protocols makes them easy to deploy in networks, and numerous industry vendors already are implementing MGCP and SIP into Voice-over-Internet Protocol (VoIP) solutions.
MGCP is central to VoIP solutions and may be integrated into products such as:
- Central Office Switches
- Gateways
- Network Access Servers
- Cable Modems
- PBXs, etc., in order to develop a convergent voice and data solution
MGCP Advantages/ Disadvantages
There are several advantages of using MGCP and IP-based communications systems over traditional telephony engineering models:
- Provides simplicity and reliability.
- Programming difficulties are concentrated on MGCs and not on the protocol.
- Service providers can develop reliable and cheap local access system.
- Provides synchronization through MGC.
- There are carrier class MGCP/Megaco media servers available today and deployed in the field. SIP lags MGCP/ Megaco in this respect.
- MGCP/Megaco is the only alternative possible today for tasks requiring signals and events, such as business conferencing or facsimile or more complex features. MGCP/Megaco's event packages are mature, tested, and deployed, whereas SIP's event packages have not yet been defined.
- Softswitches already use MGCP/ Megaco and event packages for Media Gateway control, and can reuse much of this functionality for media server control.
SIP versus MGCP:
- The SIP protocol is better specified than MGCP. Work on the MGCP protocol was distracted by the introduction of Megaco, and the MGCP specification is consequently not as solid as it might be. But although Megaco itself is better specified than MGCP, its media server events packages lag MGCP's.
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