Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[tomach <to...@dg...>]

Hello!

Yes it is .

I localized another problem. Scenario:
On sipserver i have this order of codecs:
G.711 A
G.729

On ATLSIP :
G.729
G.711 A

After SipServer send invite to ATLSIP it responses with OK and in SDP there are two codecs choosen!!! (should be only one). There is:
G. 711 A
G. 729

In ATLSIP I hear some noices all the time, i belive that codec then is uncorect (finally rtp is using g.711A I sniffed it with ethereal).

Whne I call other way: from ATLSIP to SIPSErver it works correct. always the first in ATLSIP is choosen.

Concluding I belive there are problmes when codecs are set between sipserver and atlsip. (And it seems that ATLSIP has problems wiht choosing codec). 

Scenario2:

On sipserver i have this order of codecs:
G.711 A

On ATLSIP :
G.729
G.711 A

After SipServer send invite to ATLSIP it responses with OK and in SDP there is only  one codec :
G. 711 A

But this time everything works correct. G711 A is chooosen and everythig works correct.


Did you notice this problem before?