Michael Shuler
2004-Mar-12 14:17 UTC
[Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough
When I use reinvites everything works perfectly (so phoneA<-->phoneB directly works fine). When I shut off reinvites (phoneA<-->asterisk<-->phoneB) I get the following with PhoneA initiating the call: Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference is 3360, ms is 440 Mar 12 14:43:24 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference is 2400, ms is 320 Mar 12 14:43:24 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference is 720, ms is 110 Mar 12 14:43:25 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference is 3688, ms is 481 Mar 12 14:43:25 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference is 640, ms is 100 OK, now here is the really interesting part.... When PhoneA calls PhoneB, PhoneA hears hears a jittering type of sound BUT PhoneB hears everything crystal clear. When PhoneB calls PhoneA, everything works fine...... OK, I'm stumped. ---------------------------------------- Michael Shuler, C.E.O. BitWise Systems, Inc. 1301 W. Pioneer Parkway Peoria, IL 61615 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike@bwsys.net Customer Service: (877) 976-0711
Andres
2004-Mar-12 21:42 UTC
[Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough
Michael Shuler wrote:>When I use reinvites everything works perfectly (so phoneA<-->phoneB >directly works fine). When I shut off reinvites >(phoneA<-->asterisk<-->phoneB) I get the following with PhoneA initiating >the call: > >Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference >is 3360, ms is 440 > >This is a delay issue. Packets are having greater delay than what Asterisk wants. We had the same problem. Check this BUG for a possible workaround: http://bugs.digium.com/bug_view_page.php?bug_id=0001195>Mar 12 14:43:24 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference >is 2400, ms is 320 >Mar 12 14:43:24 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference >is 720, ms is 110 >Mar 12 14:43:25 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference >is 3688, ms is 481 >Mar 12 14:43:25 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference >is 640, ms is 100 > >OK, now here is the really interesting part.... > >When PhoneA calls PhoneB, PhoneA hears hears a jittering type of sound BUT >PhoneB hears everything crystal clear. >When PhoneB calls PhoneA, everything works fine...... > >OK, I'm stumped. > >---------------------------------------- > >Michael Shuler, C.E.O. >BitWise Systems, Inc. >1301 W. Pioneer Parkway >Peoria, IL 61615 >Office: (217) 585-0357 >Cell: (309) 657-6365 >Fax: (309) 213-3500 >E-Mail: mike@bwsys.net >Customer Service: (877) 976-0711 > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Andres Network Admin http://www.telesip.net
Senad Jordanovic
2004-Mar-13 03:57 UTC
[Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough
Andres wrote:> Michael Shuler wrote: > >> When I use reinvites everything works perfectly (so phoneA<-->phoneB >> directly works fine). When I shut off reinvites >> (phoneA<-->asterisk<-->phoneB) I get the following with PhoneA >> initiating the call: >> >> Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: >> Difference is 3360, ms is 440 >> >> > This is a delay issue. Packets are having greater delay than what > Asterisk wants. We had the same problem. Check this BUG for a > possible workaround: > http://bugs.digium.com/bug_view_page.php?bug_id=0001195 >Hi, I just tried that. Did you have to recompile * in order for it to work?
Andres
2004-Mar-13 09:19 UTC
[Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough
Senad Jordanovic wrote:>Andres wrote: > > >>Michael Shuler wrote: >> >> >> >>>When I use reinvites everything works perfectly (so phoneA<-->phoneB >>>directly works fine). When I shut off reinvites >>>(phoneA<-->asterisk<-->phoneB) I get the following with PhoneA >>>initiating the call: >>> >>>Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: >>>Difference is 3360, ms is 440 >>> >>> >>> >>> >>This is a delay issue. Packets are having greater delay than what >>Asterisk wants. We had the same problem. Check this BUG for a >>possible workaround: >>http://bugs.digium.com/bug_view_page.php?bug_id=0001195 >> >> >> > >Hi, > >I just tried that. Did you have to recompile * in order for it to work? > > >Yes, you have to recompile and install again. You are changing the source code.>_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Andres Network Admin http://www.telesip.net
Senad Jordanovic
2004-Mar-14 10:07 UTC
[Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough
Olle E. Johansson wrote:> Check out the latest CVS, Mark applied changes to the code in this > area tonight. The rtp.c is changed, so the old patch in > bugs.digium.com may not be necessary any more. >Yes, it is done.. BUT Now I get MUCH higher values is the debug messages and can not understand a word from other party during the conversation. Here it is: Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference is 1386825128, ms is -1247094945 Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference is 1386825608, ms is -1247095005 Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference is 1386826088, ms is -1247095065 Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference is 1386826560, ms is -1247095124