[Freeswitch-users] wrong ptimes and second SDP!, cough cough Thomson ST2030

Alberto Escudero aep.lists at it46.se
Mon May 3 11:59:37 PDT 2010


I set in internal.xml

<param name="inbound-codec-negotiation" value="scrooge"/>

But the second SDP returns the ptime to the wrong time.

See below:
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:3596 Bah HUMBUG! Sticking
with PCMU at 8000h@20i
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:2283 Changing Codec from
PCMU at 10ms to PCMU at 20ms

--

2010-05-03 20:55:48.879237 [DEBUG] switch_core_codec.c:122
sofia/internal/1002 at 192.168.1.12:5060 Push codec L16:10
2010-05-03 20:56:38.930237 [DEBUG] sofia.c:4153 Channel
sofia/internal/1002 at 192.168.1.12:5060 entering state [received][100]
2010-05-03 20:56:38.931244 [DEBUG] sofia.c:4153 Channel
sofia/internal/1002 at 192.168.1.12:5060 entering state [completed][200]
2010-05-03 20:56:38.943283 [DEBUG] sofia.c:4153 Channel
sofia/internal/1002 at 192.168.1.12:5060 entering state [ready][200]
2010-05-03 20:56:38.943283 [DEBUG] sofia.c:4161 Duplicate SDP
v=0
o=SIP-IPPhone-0000 102406833 102406833 IN IP4 192.168.1.65
s=RTP Audio
c=IN IP4 192.168.1.65
t=0 0
m=audio 41000 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000

2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G7221:115:32000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf
send/recv payload to 97
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[telephone-event:97:8000:20]/[G7221:115:32000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G7221:107:16000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[telephone-event:97:8000:20]/[G7221:107:16000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G722:9:8000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[telephone-event:97:8000:20]/[G722:9:8000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:3596 Bah HUMBUG! Sticking
with PCMU at 8000h@20i
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:2283 Changing Codec from
PCMU at 10ms to PCMU at 20ms
2010-05-03 20:56:38.947289 [DEBUG] switch_rtp.c:1080 RE-Starting timer
[soft] 160 bytes per 20000ms
2010-05-03 20:56:38.947289 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/1002 at 192.168.1.12:5060 PCMU/8000 20 ms 160 samples
2010-05-03 20:56:38.947289 [DEBUG] switch_core_codec.c:122
sofia/internal/1002 at 192.168.1.12:5060 Push codec PCMU:0
2010-05-03 20:56:38.947289 [DEBUG] sofia_glue.c:2579 Audio params are
unchanged for sofia/internal/1002 at 192.168.1.12:5060.

-- 
Stopping junk mailers is good for the environment

> Just set codec negotiation to scrooge.
> /b
>
> On May 3, 2010, at 1:19 PM, Alberto Escudero wrote:
>
>> I  have a Thomson ST2030 VoIP phone that seams to be sending wrong
>> ptimes.
>> The SDP says 20 ms when it should be sending 10 ms.
>>
>> Freeswitch *cough cough* algorithm is able to set the soft timer for 10
>> ms... but when another SDP arrives (after approx. 30-40 seconds) the
>> ptime
>> is set back again to 20 ms.
>>
>> I am using 1.0.5 and it seems a bug in Thomson SIP stack.
>>
>> Any work around?
>> --
>> Stopping junk mailers is good for the environment
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> 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
>
>





More information about the FreeSWITCH-users mailing list