[Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Anthony Minessale
anthony.minessale at gmail.com
Mon Aug 24 12:50:57 PDT 2009
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 <mike at jerris.com> 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 PCMA at 30i,PCMU at 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/02xxxxxxxxx at fs1.my.domain) 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/02xxxxxxxxx at fs1.my.domain 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 PCMU at 30i
> >>
> >> from:
> >>
> >> <X-PRE-PROCESS cmd="set"
> >> data
> >> ="global_codec_prefs=G7221 at 32000h,G7221 at 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
> >> =G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
> >> <X-PRE-PROCESS cmd="set"
> >> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
> >>
> >>
> >> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
> >> <mailto:brian at freeswitch.org>> 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
> >> FreeSWITCH-users at lists.freeswitch.org
> >> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> 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:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> iax:guest at conference.freeswitch.org/888
> >> <http://iax:guest@conference.freeswitch.org/888>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:213-799-1400
> >> ------------------------------------------------------------------------
> >>
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> >>
> >
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>
>
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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