Paulo Vicentini wrote: > okay, > > Thanks for the tips... > > Attached are new ones...I've tested and it seems to work... > > PS: Just as i can see, my way to detect SSRC changes was > SSRC_SWITCH_MISMATCH_COUNT times of the new RTP packet stream ahead of > the current way...it was detecting first the change and already > resetting dejitter (despising transient rtp stream by "keeping ssrc > state" rather than countering new ssrc) >
I found an issue with these changes - On exit I was getting a exception on deleting the Dejitter buffer... I suggest that rather than deleting the jitter buffer, make the reset() method of the dejitter buffer public and call that. I can send diff's if you would like. Also: What is the status for a "proper" fix, i.e. the multicast / conference changes that will mix the audio ? Regards Paul _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
