Understanding the Media Gateway Peer Call Server for SRX Series Integrated Convergence Services

This topic gives an overview of the peer call server.

Under normal circumstances, the SRX Series media gateway (SRX Series MGW) relies on a SIP call server at headquarters or another location to provide it with call handling and call routing services for all SRX Series MGW analog and SIP phone endpoints at the branch. For Integrated Convergence Services, this SIP call server is referred to as a peer call server.

While the peer call server is in control, the SRX Series survivable call server (SRX Series SCS) monitors it to ensure that it is reachable. If not, the SRX Series SCS takes control and returns it to the peer call server when it is reachable again.

To configure information about the peer call server so that the SRX Series MGW can communicate with it, at a minimum you must specify the peer call server name and address.

You can specify the peer call server address in either of the following formats:

• Fully Qualified Domain Name (FQDN)
• IPv4 address

  Note: To specify multiple peer call servers, you use an FQDN. In this case, the preferred server, based on the result of the domain name services (DNS) lookup process, is used.

You can specify a description of the peer call server, for example, giving its location or type, to identify it.

You can also specify the following information for the peer call server, or accept default values if they are appropriate for your deployment.

• SIP protocol information.
  • port—The number used for the port, or communications endpoint, for the SIP protocol. The default used is the well known SIP port, 5060.
  • transport—The transport used for the SIP protocol, whether UDP, TCP, or TLS.

    The default transport is UDP.

• codecs

  One or more supported codecs specified in order of precedence in which they should be negotiated with the peer call server. If the peer call server does not support the suggested codecs, it responds with a list of codecs that it supports.

  The default set of codecs includes 711-μ, G711-A, G729AB, negotiated in that order.

• The DTMF signaling method, whether RFC 2833, inband, or sip-info.

  The default DTMF signaling method is that specified in RFC 2833.

• An authentication ID and password used by the SRX Series MGW to authenticate to the peer call server, if the peer call server requires it. You obtain this information from the peer call server administrator.
• Separate registrar address.

  An address for the registrar that is different from that of the peer call server. If you specify one, the SRX Series MGW sends REGISTER messages to the registrar address and INVITE messages to the address configured for the peer call server.

• Whether SIP registration should be disabled.

  For some SIP networks, the peer call server is aware of all media gateways. In these SIP network environments, it is not necessary for the SRX Series MGW to register to the peer call server. Were it to do so, it would introduce additional network load unnecessarily. The peer call server might silently drop the registration message and the SRX Series MGW might attempt to register to it again.

  By default, SIP registration is not disabled.

Related Topics

• Example: Configuring an Integrated Convergence Services Basic Media Gateway
• Understanding How the Media Gateway Peer Call Server Handles and Routes Calls Within the Branch for SRX Series Integrated Convergence Services
• Understanding How the Media Gateway Peer Call Server Handles and Routes Calls Outside the Branch for SRX Series Integrated Convergence Services