Hi, thanks for the answer. I finally find the problem. on the AudioMedia get_frame() method, I was setting the frame type to PJMEDIA_FRAME_TYPE_AUDIO even if there was no available frame. Setting the frame type to PJMEDIA_FRAME_TYPE_NONE when no frame are available fix my issue. 2016-03-23 19:26 GMT+01:00 Bill Gardner <billg at wavearts.com>: > Hi Simon, > > I doubt it's a problem in the bridge, more likely it's an issue with the > AudioMedia object. Try bypassing the sample processing in the AudioMedia > object and see if the leak stops. I can imagine various ways the audio can > leak through the AudioMedia object, if for example, the two instances are > using the same buffer. > > Regards, > > Bill > > > On 3/23/2016 1:49 PM, Simon Aurilac wrote: > > Sorry to insist, but I'm really don't see what I'm doing wrong. > > I have 4 conference ports attached to the bridge: > - #1 (bidirectionnel AudioMedia instance A) > - #2 (Call 0) > - #3 (bidirectionnel AudioMedia instance B) > - #4 (Call 1) > > - #0 is for the null audio device as I called pjsua_set_null_snd_dev() > > ports #1 and #2 are connected together > ports #3 and #3 are connected together > > I've checked that each ports have only 1 listener and 1 transmitter. > > But caller 1 (port #4) is earring caller 0 (port #2) even if there is no > connection established between port #4 and port #2. > > AFAIK logs seems OK: > > Call 0: > > 18:37:50.927 pjsua_call.c .....Call 0: received updated media offer > 18:37:50.927 pjsua_media.c ......Call 0: re-initializing media.. > 18:37:50.927 pjsua_media.c .......Media index 0 selected for audio call 0 > 18:37:50.927 pjsua_media.c ......Call 0: updating media.. > 18:37:50.927 pjsua_media.c ........Media stream call00:0 is destroyed > 18:37:50.927 pjsua_aud.c .......Audio channel update.. > 18:37:50.927 strm0x10480c42 ........VAD temporarily disabled > 18:37:50.927 strm0x10480c42 ........Encoder stream started > 18:37:50.927 strm0x10480c42 ........Decoder stream started > 18:37:50.927 pjsua_media.c .......Audio updated, stream #0: GSM > (sendrecv) > 18:37:50.928 pjsua_aud.c ......Conf disconnect: 2 -x- 1 > 18:37:50.928 pjsua_aud.c ......Conf disconnect: 1 -x- 2 > 18:37:50.928 pjsua_aud.c ......Conf connect: 1 --> 2 > 18:37:50.928 conference.c .......Port 1 (memspeech-0) transmitting to > port 2 (sip:nicolas at 10.2.240.109) > 18:37:50.928 pjsua_aud.c ......Conf connect: 2 --> 1 > 18:37:50.928 conference.c .......Port 2 ( <sip%3Anicolas at 10.2.240.109> > sip:nicolas at 10.2.240.109) transmitting to port 1 (memspeech-0) > 18:37:50.930 pjsua_core.c ........TX 830 bytes Response msg > 200/INVITE/cseq=104 (tdta0x104809a00) to UDP 10.2.240.109:5060: > SIP/2.0 200 OK > > Call 1: > > 18:37:59.613 pjsua_call.c .....Call 1: received updated media offer > 18:37:59.613 pjsua_media.c ......Call 1: re-initializing media.. > 18:37:59.613 pjsua_media.c .......Media index 0 selected for audio call 1 > 18:37:59.614 pjsua_media.c ......Call 1: updating media.. > 18:37:59.614 pjsua_media.c .......Call 1: stream #0 (audio) unchanged. > 18:37:59.614 pjsua_media.c .......Audio updated, stream #0: PCMU > (sendrecv) > 18:37:59.614 pjsua_aud.c ......Conf disconnect: 4 -x- 3 > 18:37:59.614 conference.c .......Port 4 ( > <sip%3Achristophe at 10.2.240.109>sip:christophe at 10.2.240.109) stop > transmitting to port 3 (memspeech-1) > 18:37:59.614 pjsua_aud.c ......Conf disconnect: 3 -x- 4 > 18:37:59.614 conference.c .......Port 3 (memspeech-1) stop transmitting > to port 4 (sip:christophe at 10.2.240.109) > 18:37:59.614 pjsua_aud.c ......Conf connect: 3 --> 4 > 18:37:59.614 conference.c .......Port 3 (memspeech-1) transmitting to > port 4 (sip:christophe at 10.2.240.109) > 18:37:59.614 pjsua_aud.c ......Conf connect: 4 --> 3 > 18:37:59.614 conference.c .......Port 4 ( > <sip%3Achristophe at 10.2.240.109>sip:christophe at 10.2.240.109) transmitting > to port 3 (memspeech-1) > 18:37:59.614 pjsua_core.c ........TX 827 bytes Response msg > 200/INVITE/cseq=104 (tdta0x10483de00) to UDP 10.2.240.109:5060: > SIP/2.0 200 OK > > Any help will be very much appreciated. > > Simon. > > > 2016-03-18 17:56 GMT+01:00 Simon Aurilac <simon.aurilac at gmail.com>: > >> OK, I made a mistake when describing this issue. >> >> Caller B is earring caller A and caller C is earring caller B (but not >> A). >> >> Don't know if it will help someone to understand this. >> >> Any help will be very appreciated. >> > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing listpjsip at lists.pjsip.orghttp://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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