[asterisk-users] One way audio problem, a=sendonly and a re-invite

[David Cunningham]

Hello all,

I have a problem where problem with one way audio, and I think it's
related to "a=sendonly" and a re-invite. Can anyone please assist?

The scenario is as follows....

- We send an INVITE to a peer, and it replies with a "100 Trying", and
then a "183 Session Progress" message containing "a=sendonly".
- Asterisk plays the caller music on hold, which I believe is correct
if we have an a=sendonly.
- Then the peer sends a "200 OK" which also has "a=sendonly", and then
sends a re-invite which I've copied and pasted below.
- We have canreinvite=no set in sip.conf, but I'm not sure if we
should be rejecting this re-invite or not because it does contain
"a=sendrecv". If it should be rejected what error should Asterisk
return, and how can we establish two way audio?

- After this re-invite Asterisk replies with a "100 Trying" and then a
"200 OK" which contains "a=recvonly".
- Call is established but called party cannot hear caller.

Here's the re-invite message - note that Asterisk is on port 5070:

U 2010/05/05 12:47:38.139701 (peer):5060 -> (asterisk):5070
INVITE sip:(called number)@(asterisk):5070 SIP/2.0.

Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594.

To: "User" <sip:(called number)@(asterisk):5070>;tag=as3ddcc528.

From: <sip:(called number)@(peer):5060>;tag=sansay7330954rdb6594.

Call-ID: 58eb52aa414c5e465c3c1a15603093fb@(asterisk).

CSeq: 2 INVITE.

Contact: <sip:(called number)@(peer):5060>.

Max-Forwards: 69.

Content-Type: application/sdp.

Content-Length: 297.

.

v=0.

o=Sansay-VSXi 188 1 IN IP4 (peer).

s=Session Controller.

c=IN IP4 (other unknown IP, maybe of called number?).

t=0 0.

m=audio 6932 RTP/AVP 18 0 8 101.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=no.

a=rtpmap:0 PCMU/8000.

a=rtpmap:8 PCMA/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-15.

a=sendrecv.

a=ptime:20.



Any help would be much appreciated!

-- 
David Cunningham, Voisonics
http://voisonics.com/
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