Hello Mathieu,
thank for your help. But this however didn't change the behaviour.
I've read of a patch in mod_sofia.c which partly corrects the problem
temporarily:
When I change Line 784 to
if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms) {
to
if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms
&& 0) {
(add a "&& 0") to deactivate this expression)
the announcements are played correctly to the Fritzbox. Connections to
other SIP phones (Snom) are also fine.
However the person at the Fritzbox still sounds very choppy in a
conference, but this is another module where I do not have a patch
available.
Best regards
Peter
Mathieu Rene schrieb:
> Try setting that in your sip profile:
>
> <param name="rtp-autofix-timing" value="false" />
>
> Thats a feature to work around with devices lying about their ptime in
> their sdp payload.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> [email protected]
>
>
>
>
> On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
>
>
>> Hello Michael,
>>
>> I made some tests with Freeswitch and Fritzbox and found by
>> Wireshark that:
>> within one call
>>
>> * Freeswitch starts sending 20msec packets, then after ~0,2 second
>> sends 30msec packets
>> * FritzBox always sends 30msec packets.
>>
>> The average jitter is below 2 msec in both directions.
>>
>> The below logs shows that Freeswitch considers the FritzBox to be
>> broken
>> and starts using 30msec packets. But there is no SIP message from FS
>> to
>> Fritzbox telling him that FB will use 30msec packets. SDP from FS to
>> Fritzbox always shows ptime:20
>>
>> BTW: We can ship you a FritzBox if you need one for testing.
>>
>> Best regards
>> Peter
>>
>>
>> Log:
>>
>> 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
>> [soft] 160 bytes per 20ms
>> 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
>> v=0
>> o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
>> s=FreeSWITCH
>> c=IN IP4 182.xxx.xx.xxx
>> t=0 0
>> m=audio 30290 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
>> sofia/internal/[email protected]!
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send
>> signal
>> sofia/internal/[email protected] [BREAK]
>> EXECUTE sofia/internal/[email protected]
>> playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated l...@8000hz 1 channels 20ms
>> 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
>> sofia/internal/[email protected] entering state [early][183]
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
>> sofia/internal/[email protected] receive message
>> [TRANSCODING_NECESSARY]
>> 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to
>> use
>> ptime 20 but what they meant to say was 30
>> This issue has so far been identified to happen on the following
>> broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so
>> broken
>> who knows what will happen..
>>
>>
>>
>>
>>
>>
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>
>
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