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 _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
