[OpenSIPS-Users] Music on hold, sdp 0.0.0.0.

[Răzvan Crainea]

Unfortunately the trace is no longer there. Can you please repost it?

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 06/20/2013 01:23 PM, Miha wrote:
> HI Razvan,
>
>   here is a trace: http://pastebin.com/QgUrJTwi
>
> A calls B, B make 2-trasfer to C. (first A hears B, after 2-transfer is
> made A is put on hold and it hears music). When transfer is made, A and
> C are compined but A still hears music on hold, it should be speaking to C.
>
> Thanks!
>
> miha
>
> Dne 6/20/2013 12:24 PM, piše Răzvan Crainea:
>> Hi, Miha!
>>
>> Indeed, this is a classical late-negociation scenario, when the media
>> session is established by the 200OK and the ACK.
>> However, I am not sure what you are trying to achieve. So you are
>> saying that B puts A on hold and then transfers the call to C. But
>> what happens to the B leg? I don't really understand why A should
>> still be able to talk to B.
>> Also, can A and C hear each other? I am not sure what is the desired
>> behavior. Can you please give a little more details about the call?
>> Probably a SIP trace will be helpful.
>>
>> Best regards,
>>
>> Razvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 06/20/2013 08:50 AM, Miha wrote:
>>> Dne 6/20/2013 8:32 AM, piše Miha:
>>>> Hi,
>>>>
>>>> I need one info regarding puting user on hold.
>>>>
>>>> A calls B, B puts A on hold and do 2-transfer to C. After C answers it
>>>> puts A and C together.
>>>>
>>>> A is on our side, using opensips and FS for media gw, B and C on other
>>>> side using netcetrex softswitch.
>>>>
>>>> A sends invite with SDP to B, B sends OK, and send reinvite with SDP
>>>> c=0.0.0.0. which is ok, A sends ok and music is heard by A, after B
>>>> trasfer call to C and C pickup call, A gets invite buth with no SDP.
>>>> is this ok? I guess it should sends invite with sdp which has IP for
>>>> sending RTP. This results that A and B can not hear each other.
>>>>
>>>> Other side is saying that with other voip providers this works ok.
>>>>
>>>> thanks!
>>>>
>>>> Miha
>>>>
>>>> _______________________________________________
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>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>> I also noticed one thing. After they send invite without SDP, we sends
>>> SDP 200 ok and they send back ACK SDP in which they send c=IP. According
>>> to RFC this is right way.
>>> What I am missing here.
>>>
>>> thanks!
>>>
>>> miha
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> _______________________________________________
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>