[asterisk-users] No rtpmap codec info in 200 OK

[Bruce B]

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.

-Bruce

On Mon, Dec 19, 2011 at 7:41 PM, William Scott <william at magicwilly.info>wrote:

> > It seems quite unlikely that the presence of
> > an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have
> any
> > problems.
>
> Thanks for the reply.
>
> I'll expand on the scenario...
>
> This particular ATA does not send  'a=rtpmap' for any codec.
>
> When talking to a Asterisk PBX everything works fine.
>
> When talking to a VSP that sends an INVITE with "User-Agent: Sippy"
> the call is setup then drops after 32 seconds.
>
> Packet captures shows that no ACK is received after the ATA sends the
> 200 OK (missing rtpmap). After sending 200 OK about 6 times it then
> sends BYE and the call disconnects.
>
> Every other ATA I have sends rtpmap and works fine.
>
> The idea was to manipulate Asterisk into not sending rtpmap for the
> codec to confirm what happens.
>
> I'll now look for another solution.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111219/7c5b8f0c/attachment.htm>