Yeah, I see. I have to fix this. On Wed, Oct 29, 2008 at 12:46 AM, Paulo Vicentini <[EMAIL PROTECTED]> wrote: > 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 > >
-- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000 _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
