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1-way audio after re-invite
Posted: 13 Jan 2010 21:29
by kishanj
Hi,
I'm able to establish a call from a SIP extension to a SIP trunk. At this point, there is 2-way audio. A few seconds after the call is established, the SIP extension modifies the SDP (connection parameters - c line) via re-invite. OXE sends a 200 OK with the SDP of the remote party but doesn't renegotiate the new SDP with the other (trunk) leg. As a result, the endpoint on the other side of the trunk continues to send RTP packets to the connection negotiated in the initial INVITE.
Can someone help me understand why OXE wouldn't re-invite the trunk leg? traced snapshot attached.
Thanks.
Posted: 20 Jan 2010 23:18
by frank
Hi ..
I'm a little bit lazy here.. I can see 4 calls.. Can you make 1 trace only with 1 call ?
Thanks
Posted: 21 Jan 2010 03:01
by freedom
We also had problems with one-way communications between sip device and sip trunk, but then with the call setup.
We had to change the SDP in 18x parameter to fix it.
Check the setting of 'SDP in 18x' in the sip gateway (and external sip gateway if you use this).
By default this is set to TRUE, but some applications are not able to use this.
Change it to FALSE and see if this changes anything.
Posted: 21 Jan 2010 13:09
by kishanj
frank wrote:Hi ..
I'm a little bit lazy here.. I can see 4 calls.. Can you make 1 trace only with 1 call ?
Thanks
The very 1st call (Call-ID: 8372f800-b4e15e19-1681fc-8601a8c0@192.168.1.134) is rejected with a 422 Response.
The 2nd call (Call-ID: 840b8e80-b4e15e1a-1681fd-8601a8c0@192.168.1.134) is redirected.
You can ignore these 2 calls. Sorry about that!
The rest of the log has 2 legs of 1 call.
The 1st leg is between OXE & SIP Extension (Call-ID: b74e730-6f03a8c0-13c4-50030-1d40b8-88420b3-1d40b8)
The 2nd leg is between OXE & Sip Trunk (Call-ID: 57f4b2038ed31de2c21a38fbfcbd8af8@192.168.7.200).
OXE receives the call from SIP Extension and routes the call to the sip trunk based on the dialed number.
OXE: 192.168.7.200:5060
SIP Extension: 192.168.3.111:5060
SIP Trunk: 192.168.1.134:5068
Posted: 21 Jan 2010 13:10
by kishanj
freedom wrote:We also had problems with one-way communications between sip device and sip trunk, but then with the call setup.
We had to change the SDP in 18x parameter to fix it.
Check the setting of 'SDP in 18x' in the sip gateway (and external sip gateway if you use this).
By default this is set to TRUE, but some applications are not able to use this.
Change it to FALSE and see if this changes anything.
It is already set to FALSE.
Re: 1-way audio after re-invite
Posted: 13 Jan 2016 04:22
by thanzeel
Hi all,
I have the same issue. I have tried toggling the values but still the same behaviour. Has anyone got a solution for this.