[Asterisk-Users] * and Verisign SIP-7 service

[Kevin P. Fleming]

Matthew Crocker wrote:

> The info sheet at Verisign says their SIP-7 product does the MGCP with 
> my AS5400 and SIP with a sip server (Asterisk).  An inbound call would 
> generate a SS7 ISUP message to Verisign.  They would send a SIP message 
> to Asterisk.  Asterisk would respond with a SIP message back pointing to 
> a 'sip phone'  Verisign would convert the 'sip phone' to a MGCP 
> server/port and send the MGCP message to my gateway.

That's very close, I believe. Actually what will happen is that the 
SIP-7 server will accept the ISUP call initiation, talk to the gateway 
via MGCP to allocate a trunk port and get an RTP IP/port number pair, 
then send a SIP INVITE to Asterisk.

This INVITE will come from the SIP-7 server, but the embedded SDP data 
will show that the media stream (RTP) will be coming from the gateway. 
If Asterisk accepts the SIP call, then it will respond OK, which will 
cause the SIP-7 server to accept the ISUP call, nail up the trunk port 
in the gateway, and let the call proceed.

While the call is in process, SIP traffic will continue to go between 
the SIP-7 server and the Asterisk server, but the RTP will flow directly 
between the gateway and the Asterisk server.