ptime problem with provider (Cisco hardware)
Hi, I have some troubles with provider configuration. The are warnings in logs: 2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 40 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.. My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x). Moreover, above warning appears only when someone calls to FreeSWITCH (voice quality is poor) - connection from FreeSWITCH works without any warnings and has perfect quality. Other my providers works perfect. Is there any solution of this problem? regards, Lukasz
Hi, This is a known issue with some of these platforms but for completeness can you send the actual SDP? 2009/3/19 Łukasz Czerpak <lukasz@czerpak.eu>:
Hi,
I have some troubles with provider configuration. The are warnings in logs:
2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 40
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..
My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x).
Moreover, above warning appears only when someone calls to FreeSWITCH (voice quality is poor) - connection from FreeSWITCH works without any warnings and has perfect quality.
Other my providers works perfect.
Is there any solution of this problem?
regards, Lukasz
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-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
Kristian Kielhofner wrote:
Hi,
This is a known issue with some of these platforms but for completeness can you send the actual SDP?
My voip configuration: +------------+ +----------+ | ROUTER | LAN | Linksys | NET----| with |---------| PAP2T-EU | | FreeSWITCH | | | +------------+ +----------+ Linksys has changed rtp packet size to 0.020 On FreeSWITCH i am using g729 (http://freehg.org/u/deepwalker/fs_g729/) - for testing only. SDP logs are below, full session's log is in attachment: 2009-03-19 20:07:02 [DEBUG] sofia.c:2806 sofia_handle_sip_i_state() Channel sofia/external/0607xxxxxx@217.11.128.50 entering state [received] 2009-03-19 20:07:02 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 9221 3981 IN IP4 217.11.128.50 s=SIP Call c=IN IP4 217.11.128.50 t=0 0 m=audio 18292 RTP/AVP 18 4 2 98 99 0 8 3 100 101 19 c=IN IP4 217.11.128.50 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:2 G726-32/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:99 G726-16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=direction:active ... 2009-03-19 20:07:02 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237472006 1237472007 IN IP4 89.79.191.29 s=FreeSWITCH c=IN IP4 89.79.191.29 t=0 0 m=audio 17616 RTP/AVP 18 101 19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 a=sendrecv ... 2009-03-19 20:07:03 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 121150237 121150237 IN IP4 192.168.1.250 s=- c=IN IP4 192.168.1.250 t=0 0 m=audio 16438 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 ... 2009-03-19 20:07:03 [DEBUG] mod_sofia.c:503 sofia_answer_channel() Local SDP sofia/external/0607320038@217.11.128.50: v=0 o=FreeSWITCH 1237472006 1237472008 IN IP4 89.79.191.29 s=FreeSWITCH c=IN IP4 89.79.191.29 t=0 0 m=audio 17616 RTP/AVP 18 101 19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 a=sendrecv ... 2009-03-19 20:07:04 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 9221 3981 IN IP4 217.11.128.50 s=SIP Call c=IN IP4 217.11.128.50 t=0 0 m=audio 18292 RTP/AVP 18 19 101 100 c=IN IP4 217.11.128.50 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=ptime:60 regards, Lukasz
Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729@60i and see what it does. /b On Mar 19, 2009, at 2:20 PM, Łukasz Czerpak wrote:
Linksys has changed rtp packet size to 0.020 On FreeSWITCH i am using g729 (http://freehg.org/u/deepwalker/ fs_g729/) - for testing only.
SDP logs are below, full session's log is in attachment:
Brian West wrote:
Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729@60i and see what it does.
I've just tested G729@60i and everything works perfect - thank you very much. I didn't test ulaw. What is wrong - my provider is incompatible with specification or FreeSWITCH has problem with codec negotiation? Is there any possibility to force codec for specific gateway/provider? regards Lukasz
The issue I seen was they invite to you with NO ptime which indicates 20ms, they should invite with ptime:60 if they want 60. /b On Mar 19, 2009, at 3:06 PM, Łukasz Czerpak wrote:
I've just tested G729@60i and everything works perfect - thank you very much. I didn't test ulaw. What is wrong - my provider is incompatible with specification or FreeSWITCH has problem with codec negotiation?
Is there any possibility to force codec for specific gateway/provider?
regards Lukasz
Brian West wrote:
The issue I seen was they invite to you with NO ptime which indicates 20ms, they should invite with ptime:60 if they want 60.
I see but there is any solution to bypass this provider's "incompatibility"? I want to stay with this provider anyway - he has very good quality and nice prices ;) thanks and regards, Łukasz
Well you said you were using G.729 for testing... when you're clearly not... but I told you already how to fix it... for that IP or peer G729@60i /b On Mar 19, 2009, at 4:03 PM, Łukasz Czerpak wrote:
I see but there is any solution to bypass this provider's "incompatibility"? I want to stay with this provider anyway - he has very good quality and nice prices ;)
thanks and regards, Łukasz
Brian West pisze:
Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729@60i and see what it does.
I made some tests with ulaw with success. There is no problem with ptime negotiation. Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? regards, -- Łukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net]
I am not a Cisco expert, but as far as I know packetization period is configurable in Cisco. When you specify the codec for a dial peer, you can set up the ptime value in milliseconds. This is an optional argument in the Cisco command line and what happens when providers don't specify it is that your end assumes a convenient or default ptime value, that probably don't match with far end default... A solution would be to figure out what is the convenient value for an individual provider and set up FS to match it, or suggest your provider to specify a ptime in the first request. By the way, Cisco supports 10, 20, 30, 40, 50 and 60 as ptime values for codecs G.729, G.729A, G.729B and G.729AB. Pablo 2009/3/20 Łukasz Czerpak <lukasz@czerpak.eu>:
Brian West pisze:
Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729@60i and see what it does.
I made some tests with ulaw with success. There is no problem with ptime negotiation.
Will the g729 codec be fully (not passthrough) supported in FreeSWITCH?
regards,
-- Łukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net]
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The pvalue of the originator overrides whatever you might have configured on Cisco GW. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pablo Hernan Saro Sent: 21 March 2009 03:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] ptime problem with provider (Cisco hardware) I am not a Cisco expert, but as far as I know packetization period is configurable in Cisco. When you specify the codec for a dial peer, you can set up the ptime value in milliseconds. This is an optional argument in the Cisco command line and what happens when providers don't specify it is that your end assumes a convenient or default ptime value, that probably don't match with far end default... A solution would be to figure out what is the convenient value for an individual provider and set up FS to match it, or suggest your provider to specify a ptime in the first request. By the way, Cisco supports 10, 20, 30, 40, 50 and 60 as ptime values for codecs G.729, G.729A, G.729B and G.729AB. Pablo 2009/3/20 Łukasz Czerpak <lukasz@czerpak.eu>:
Brian West pisze:
Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729@60i and see what it does.
I made some tests with ulaw with success. There is no problem with ptime negotiation.
Will the g729 codec be fully (not passthrough) supported in FreeSWITCH?
regards,
-- Łukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net]
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
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participants (5)
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Abdul Hakeem
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Brian West
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Kristian Kielhofner
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Pablo Hernan Saro
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Łukasz Czerpak