mr fritz is lying somewhere get a pcap of the traffic from fritz to FS and look at the size of the audio packets if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252) then it's 30ms
if it's saying 20 but it means 30 you should leave the last change in place and also add in <param name="rtp-autofix-timing" value="true" /> and <param name="inbound-codec-negotiation" value="scrooge"/> to overcome the bug on their end. On Mon, Aug 24, 2009 at 2:05 PM, Michael Jerris <[email protected]> wrote: > That is the remote sdp, not the local sdp. They are sending ptime 20, > not us. Are they actually sending 20 ms packets or are they sending 30? > > MIke > > On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote: > > > Hello Anthony, > > > > I set p...@30i,p...@30i and I can see in the logs that PCMA is used. > > However ptime is set to 20 msec as shown in the Logs: > > > > 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP: > > v=0 > > o=user 2075230 2075230 IN IP4 217.xx.xx.xxx > > s=call > > c=IN IP4 217.xx.xx.xxx > > t=0 0 > > m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101 > > a=rtpmap:2 G726-32/8000 > > a=rtpmap:102 G726-32/8000 > > a=rtpmap:100 G726-40/8000 > > a=rtpmap:99 G726-24/8000 > > a=rtpmap:97 iLBC/8000 > > a=fmtp:97 mode=30 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-11 > > a=rtcp:7079 > > > > 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404 > > (sofia/internal/[email protected]) State NEW > > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec > > Compare > > [PCMA:8:8000:0]/[G722:9:8000:20] > > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec > > Compare > > [PCMA:8:8000:0]/[PCMU:0:8000:20] > > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec > > Compare > > [PCMA:8:8000:0]/[PCMA:8:8000:20] > > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec > > sofia/internal/[email protected] PCMA/8000 20 ms 160 samples > > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf > > payload to 101 > > > > Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is > > horrible. > > > > Best regards > > Peter > > > > Anthony Minessale schrieb: > >> try setting FS to 30ms too > >> > >> edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it > >> looks like p...@30i > >> > >> from: > >> > >> <X-PRE-PROCESS cmd="set" > >> data > >> ="global_codec_prefs=g7...@32000h,g7...@16000h,G722,PCMU,PCMA,GSM"/> > >> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> > >> > >> to: > >> > >> <X-PRE-PROCESS cmd="set" > >> data > >> = > >> "global_codec_prefs > >> =g7...@32000h,g7...@16000h,G722,p...@30i,p...@30i,GSM"/> > >> <X-PRE-PROCESS cmd="set" > >> data="outbound_codec_prefs=p...@30i,p...@30i,GSM"/> > >> > >> > >> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <[email protected] > >> <mailto:[email protected]>> wrote: > >> > >> You can ship me one whois bkw.org <http://bkw.org>, I can add it > >> to my lab. > >> > >> /b > >> > >> On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote: > >> > >>> > >>> BTW: We can ship you a FritzBox if you need one for testing. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> [email protected] > >> <mailto:[email protected]> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ > >> freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:[email protected] <msn%[email protected]> > >> <mailto:msn%[email protected]<msn%[email protected]> > > > >> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > >> <mailto:paypal%[email protected]<paypal%[email protected]> > > > >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:[email protected] <sip%[email protected]> > >> <mailto:sip%[email protected]<sip%[email protected]> > > > >> iax:[email protected]/888 > >> <http://iax:[email protected]/888> > >> googletalk:[email protected]<googletalk%3aconf%[email protected]> > >> <mailto:googletalk%3aconf%[email protected]<googletalk%253aconf%[email protected]> > > > >> pstn:213-799-1400 > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> [email protected] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > [email protected] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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