wrong ptimes and second SDP!, cough cough Thomson ST2030
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
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
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@8000h@20i 2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:2283 Changing Codec from PCMU@10ms to PCMU@20ms -- 2010-05-03 20:55:48.879237 [DEBUG] switch_core_codec.c:122 sofia/internal/1002@192.168.1.12:5060 Push codec L16:10 2010-05-03 20:56:38.930237 [DEBUG] sofia.c:4153 Channel sofia/internal/1002@192.168.1.12:5060 entering state [received][100] 2010-05-03 20:56:38.931244 [DEBUG] sofia.c:4153 Channel sofia/internal/1002@192.168.1.12:5060 entering state [completed][200] 2010-05-03 20:56:38.943283 [DEBUG] sofia.c:4153 Channel sofia/internal/1002@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@8000h@20i 2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:2283 Changing Codec from PCMU@10ms to PCMU@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@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@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@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
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I managed to get it working by forcing the phone to use G.711 with 20 ms and not 10 ms. But for the sake of understanding what exactly <param name="inbound-codec-negotiation" value="scrooge"/> and <param name="rtp-autofix-timing" value="true"/> does? I saw a BUG in Jira the refers to this second SDP for Sonus http://jira.freeswitch.org/browse/FSRTP-8 I wonder if this is Sonus specific patch. Unfortunately we do not have 16 oz Hammers here in Sweden... so i recommend this alternative! http://www.ikea.com/gb/en/catalog/products/70082653 -- Stopping junk mailers is good for the environment
Might I recommend the Craftsman 16 oz. Rip Claw Hammer?
/b
On May 3, 2010, at 1:59 PM, Alberto Escudero wrote:
I set in internal.xml
<param name="inbound-codec-negotiation" value="scrooge"/>
But the second SDP returns the ptime to the wrong time.
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Hi, have you tried an other firmware version. You could find firmware there: http://www.thomsonbroadbandpartner.com/telephony-solutions/products/product-... rod Le 04/05/2010 00:55, Alberto Escudero a écrit :
I managed to get it working by forcing the phone to use G.711 with 20 ms and not 10 ms. But for the sake of understanding what exactly
<param name="inbound-codec-negotiation" value="scrooge"/> and <param name="rtp-autofix-timing" value="true"/> does?
I saw a BUG in Jira the refers to this second SDP for Sonus http://jira.freeswitch.org/browse/FSRTP-8 I wonder if this is Sonus specific patch.
Unfortunately we do not have 16 oz Hammers here in Sweden... so i recommend this alternative! http://www.ikea.com/gb/en/catalog/products/70082653
participants (3)
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Alberto Escudero
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Brian West
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Rod.