Hi, What version of SFLphone are you running? Is this problem occurs between two SFLphone instances? Is the stream audio stopped in both way (incoming/outgoing), or in only one way?
When putting the phone on hold and taking it back, you reinitialize the RTP session and thus reinitialize the RTP local timestamp. I know that our RTP stack will, by default, consider late RTP packets as expired and thus drop them. So my first guess would be that the RTP stack is not processing fast enough the incoming data packets. On way to solve this would be to implement the OST::RtpSession::onPacketReceive method callback in src/audio/audiortp/AudioRtpSession.h . This may require some refactoring of the AudioRtpFactory as it use templates, which change the callback function signature. Let me know! Regards, Alexandre > ----- Mail Original ----- > De: "Jan Bakuwel" <[email protected]> > À: [email protected] > Envoyé: Vendredi 22 Octobre 2010 19h13:18 GMT -05:00 USA/Canada - États de > l'Est > Objet: [SFLphone] Debugging SFLphone > > Hi, > > Is there a way to "end-user" debug SFLphone? It works really well for me > but sometimes (ie not always?) there's no audio on a call just initiated > until I put the call on hold then retrieve it again. I can live with the > workaround ... but my conversation partners will be wondering where I am > those first few seconds of the call. > > Jan _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
