I could be wrong but this sounds like a NAT issue rather SIP related packet issue. You are not receiving a response back is what I get a lot of times when my NAT is not setup properly. Call goes on for 10 or 20 second (I try the echo application and it hangs up before I get to talk) and then cuts off.<div>
<br></div><div>-Bruce<br><br><div class="gmail_quote">On Mon, Dec 19, 2011 at 7:41 PM, William Scott <span dir="ltr"><<a href="mailto:william@magicwilly.info">william@magicwilly.info</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
> It seems quite unlikely that the presence of<br>
> an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any<br>
> problems.<br>
<br>
Thanks for the reply.<br>
<br>
I'll expand on the scenario...<br>
<br>
This particular ATA does not send 'a=rtpmap' for any codec.<br>
<br>
When talking to a Asterisk PBX everything works fine.<br>
<br>
When talking to a VSP that sends an INVITE with "User-Agent: Sippy"<br>
the call is setup then drops after 32 seconds.<br>
<br>
Packet captures shows that no ACK is received after the ATA sends the<br>
200 OK (missing rtpmap). After sending 200 OK about 6 times it then<br>
sends BYE and the call disconnects.<br>
<br>
Every other ATA I have sends rtpmap and works fine.<br>
<br>
The idea was to manipulate Asterisk into not sending rtpmap for the<br>
codec to confirm what happens.<br>
<br>
I'll now look for another solution.<br>
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