Yeah, DEcoder is considering the same stream , but stream has changed: if (mIsStreamInitialized == FALSE) ---> use previous mStreamState...but we have other one
Paulo On Tue, Oct 28, 2008 at 5:37 PM, Alexander Chemeris < [EMAIL PROTECTED]> wrote: > Hum, if FromNet correctly sets preferred SSRC, then it should be a problem > with Decoder I think. > > On Tue, Oct 28, 2008 at 10:49 PM, Paulo Vicentini > <[EMAIL PROTECTED]> wrote: > > "Did you try to decode and replay from the Statistics / voip menu to see > if > > RTP contains sound or silence ?" > > > > Yes, captured RTP packet contains sound from remote EP (decoded with > > wireshark ) > > > > it seems that there is a problem when changing stream ID: > > > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x42DFFDD0 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x1C699DD2 > > pushPacket: Pref:0x1C699DD2 > > > > pushPacket: Pref:0x1C699DD2 > > > > On Tue, Oct 28, 2008 at 3:31 PM, stipus <[EMAIL PROTECTED]> wrote: > >> > >> Did you try to decode and replay from the Statistics / voip menu to see > if > >> RTP contains sound or silence ? > >> > >> > >> > >> > >> > >> > >> > >> De : Paulo Vicentini [mailto:[EMAIL PROTECTED] > >> Envoyé : mardi 28 octobre 2008 19:16 > >> À : stipus > >> > >> Objet : Re: [sipxtapi-dev] stop to playback decoded RTP > >> > >> > >> > >> capture2.rar > >> > >> On Tue, Oct 28, 2008 at 11:27 AM, stipus <[EMAIL PROTECTED]> wrote: > >> > >> As long as you didn't use Wireshark to capture and replay the RTP, you > >> can't be sure that this RTP doesn't contain silence…. > >> > >> > >> > >> It's a 10 minute test (time to download and install wireshark and > capture > >> a few packets), and then you can be sure if it's a problem within > sipxtapi > >> or not. > >> > >> > >> > >> stipus > >> > >> > >> > >> > >> > >> De : [EMAIL PROTECTED] > >> [mailto:[EMAIL PROTECTED] De la part de Paulo > >> Vicentini > >> Envoyé : mardi 28 octobre 2008 14:52 > >> > >> À : [email protected] > >> > >> Objet : Re: [sipxtapi-dev] stop to playback decoded RTP > >> > >> > >> > >> Hi, > >> After the end of the announcement, valid RTP packets are entering thru > >> NetInTask (get1Msg) and apparently they are pushed to the queue, so that > it > >> is not a network issue. > >> > >> The strange thing, although I can't hear anything from remote, is that > >> SpkrThread still receives messages from Queue (pMsg = 0x00bc12b8 > >> {mpData1=957 mpData2=1430840 }) and waveOutWrite is being called without > >> error ( returning 0) > >> > >> tks > >> > >> Paulo > >> > >> > >> > >> On Mon, Oct 27, 2008 at 6:44 PM, stipus <[EMAIL PROTECTED]> wrote: > >> > >> You can use Wireshark to capture all IP packets, and then decode & play > >> received and sent RTP (if it's in one of the supported formats – only > G711 > >> as far as I know). > >> > >> > >> > >> You can access the RTP replay feature from the wireshark statistics / > VOIP > >> calls menu. > >> > >> > >> > >> stipus > >> > >> > >> > >> De : [EMAIL PROTECTED] > >> [mailto:[EMAIL PROTECTED] De la part de Paulo > >> Vicentini > >> Envoyé : lundi 27 octobre 2008 20:53 > >> À : [email protected] > >> Objet : [sipxtapi-dev] stop to playback decoded RTP > >> > >> > >> > >> I am facing a problem in a specific scenario: > >> > >> sipxtapiEP----------------media server plays > message-------------------EP > >> (several kinds) > >> > >> (both EPs are able to hear the message) > >> > >> sipxtapiEP(can't hear other EP ) ------------(message is > >> over)------------------EP (is able to hear sipxtapiEP) > >> > >> Whenever an ingoing / outgoing call is intercepted by a media server > that > >> plays a message (e.g announcement / collect call message) before > bridging > >> between the finals endpoints I have the following problem: > >> While playing the message both endpoints are able to hear the message > but > >> after all the message is played, sipxtapi EP is mute (can't hear other > end > >> EP), although it is able to send / receive RTP. > >> The other end is able to hear (audio voice) sipxtapi EP after message is > >> played. > >> SDP session description remains the same by the end of the announcements > >> (no re-invite). > >> Without a media server (announcement) in mid of the call all is fine > >> (receive / make). > >> > >> I do check NetInTask / MprFromNet and I saw that RTP packets are still > >> been pushed after the end of the message. > >> I will check Spk Task and others to see what's going on…but any help to > >> figure out my issue is welcome. > >> > >> Thank you > >> > >> > >> _______________________________________________ > >> sipxtapi-dev mailing list > >> [email protected] > >> List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/ > >> > >> > >> > >> _______________________________________________ > >> sipxtapi-dev mailing list > >> [email protected] > >> List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/ > >> > >> > > > > _______________________________________________ > > sipxtapi-dev mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/ > > > > > > -- > Regards, > Alexander Chemeris. > > SIPez LLC. > SIP VoIP, IM and Presence Consulting > http://www.SIPez.com > tel: +1 (617) 273-4000 >
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