forcing ptime settings
I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem. but now I'm getting a [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 error and in some sessions the audio is playing back too fast (at 3x the speed). Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds? --matt
That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff. /b On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH.
The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20
I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem.
but now I'm getting a
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10
error and in some sessions the audio is playing back too fast (at 3x the speed).
Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds?
--matt
They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes. On Wed, Dec 23, 2009 at 9:55 AM, Brian West <brian@freeswitch.org> wrote:
That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH.
The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20
I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem.
but now I'm getting a
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10
error and in some sessions the audio is playing back too fast (at 3x the speed).
Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds?
--matt
_______________________________________________ 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-users http://www.freeswitch.org
-- -Rupa
If I only care about outbound audio, is there a way to force the audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there still this same issue? --matt On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker <rupa@rupa.com> wrote:
They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes.
On Wed, Dec 23, 2009 at 9:55 AM, Brian West <brian@freeswitch.org> wrote:
That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH.
The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20
I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem.
but now I'm getting a
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10
error and in some sessions the audio is playing back too fast (at 3x the speed).
Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds?
--matt
_______________________________________________ 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-users http://www.freeswitch.org
-- -Rupa
_______________________________________________ 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-users http://www.freeswitch.org
You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: <param name="rtp-autofix-timing" value="false" /> Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 23-Dec-09, at 11:41 AM, Matthew Fong wrote:
If I only care about outbound audio, is there a way to force the audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there still this same issue?
--matt
On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker <rupa@rupa.com> wrote: They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes.
On Wed, Dec 23, 2009 at 9:55 AM, Brian West <brian@freeswitch.org> wrote: That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH.
The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20
I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem.
but now I'm getting a
[WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10
error and in some sessions the audio is playing back too fast (at 3x the speed).
Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds?
--matt
_______________________________________________ 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-users http://www.freeswitch.org
-- -Rupa
_______________________________________________ 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-users http://www.freeswitch.org
_______________________________________________ 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-users http://www.freeswitch.org
You might also have to set the codec negotiation to scrooge /b On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote:
You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty:
<param name="rtp-autofix-timing" value="false" />
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
Setting the codec negotiation to scrooge resolved my problems w/ CallCentric. I'd bet that'd do it for him as well. Lessons Learned by me: 1.) Listen to Brian. 2.) When in doubt, refer to rule 1. Best Regards, Karl J. Vesterling kjv@ken-ton.com 202-461-3231 x0 On Dec 23, 2009, at 11:57 AM, Brian West wrote:
You might also have to set the codec negotiation to scrooge
/b
On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote:
You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty:
<param name="rtp-autofix-timing" value="false" />
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
_______________________________________________ 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-users http://www.freeswitch.org
On Sun, Dec 27, 2009 at 5:01 AM, Karl J. Vesterling <kjv@ken-ton.com> wrote:
Setting the codec negotiation to scrooge resolved my problems w/ CallCentric.
I'd bet that'd do it for him as well.
*Lessons Learned by me:* 1.) Listen to Brian. 2.) When in doubt, refer to rule 1.
Can I get that framed? :) -MC
participants (6)
-
Brian West
-
Karl J. Vesterling
-
Mathieu Rene
-
Matthew Fong
-
Michael Collins
-
Rupa Schomaker