Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg
Hi, I have very strange problem with audio after upgrading of FS to the newest GIT version. Previous version was quite old, but worked without problems for a long time: FreeSWITCH version: 1.0.head (git-313b164 2011-11-26 08-53-01 -0600) Current version is: FreeSWITCH version: 1.5.1b () All configuration files are the same (exact copy) System: Debian Lenny 32-bit All calls are made over SIP Symptoms: - IVR prompt playback is OK - FAX sending/receiving is OK - Calling to remote servers via VoIP provider is OK - The only problem is calling to local user and it's very strange. When external user calls local user (SIP call, via IVR, then bridge), audio stream from external user is OK, but audio stream from local user sent to external user is very delayed. Delay increases during a call. At the beginning is about 1 second, all the time delay is increasing. Audio is reaching external user but after about 30 seconds of call stream is delayed for about 10 seconds. At the same time audio stream from external user is heard by local user with no delays. Propably something changed in FS core or config files and I need to adjust to the new version. As I stated before configuration is exact copy and worked perfectly on older version. Any hints what should I check or change to solve this problem? Kind Regards
Additional info from FS console: 2013-05-23 10:30:40.402324 [DEBUG] mod_sofia.c:642 SOFIA EXCHANGE_MEDIA 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/myusername@my.office_pbx.ip [BREAK] 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/caller_number@my.fs.ip [BREAK] 2013-05-23 10:30:40.422955 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 480 bytes to accommodate 320->480 2013-05-23 10:30:40.450958 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 320 bytes to accommodate 320->320 2013-05-23 10:30:40.662295 [WARNING] switch_core_media.c:1370 Asynchronous PTIME not supported, changing our end from 0 to 20 2013-05-23 10:30:40.662295 [DEBUG] switch_core_media.c:1741 Already using PCMA
Probably best to open a bug on jira.freeswitch.org. On 23/05/13 09:40, Adam Raszynski wrote:
Additional info from FS console:
2013-05-23 10:30:40.402324 [DEBUG] mod_sofia.c:642 SOFIA EXCHANGE_MEDIA 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/myusername@my.office_pbx.ip [BREAK] 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/caller_number@my.fs.ip [BREAK] 2013-05-23 10:30:40.422955 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 480 bytes to accommodate 320->480 2013-05-23 10:30:40.450958 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 320 bytes to accommodate 320->320 2013-05-23 10:30:40.662295 [WARNING] switch_core_media.c:1370 Asynchronous PTIME not supported, changing our end from 0 to 20 2013-05-23 10:30:40.662295 [DEBUG] switch_core_media.c:1741 Already using PCMA
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I wonder if this is related to your Asynchronous PTIME ??? are you doing 20ms in one direction and 30 the other ?? On 23 May 2013 18:40, Adam Raszynski <netcentrica@gmail.com> wrote:
Additional info from FS console:
2013-05-23 10:30:40.402324 [DEBUG] mod_sofia.c:642 SOFIA EXCHANGE_MEDIA 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/myusername@my.office_pbx.ip [BREAK] 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/caller_number@my.fs.ip [BREAK] 2013-05-23 10:30:40.422955 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 480 bytes to accommodate 320->480 2013-05-23 10:30:40.450958 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 320 bytes to accommodate 320->320 2013-05-23 10:30:40.662295 [WARNING] switch_core_media.c:1370 Asynchronous PTIME not supported, changing our end from 0 to 20 2013-05-23 10:30:40.662295 [DEBUG] switch_core_media.c:1741 Already using PCMA
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-- Sincerely Jay
I use default sofia settings. I'm not sure how can I "do" or not to do 20 or 30ms - what params are responsible for that? For the calling party I have no control - call comes from external provider, but I can change this for local user. What params should I change? 2013/5/23 jay binks <jaybinks@gmail.com>
I wonder if this is related to your Asynchronous PTIME ??? are you doing 20ms in one direction and 30 the other ??
I changed outbound-codec-prefs to PCMU@20i,PCMA@20i so timing on inbound and outbound legs is the same Info about async PTIME does not appear now in console, so I think now there are synchronous But the problem is not solved - audio stream from local user is still lagging
Changed also rtp-timer-name to none now bridged call is OK, but when local user uses IVR provided by bind-meta-app it gets distorted audio, when leaving IVR and switching back to the caller audio is OK again 2013/5/23 Adam Raszynski <netcentrica@gmail.com>
I changed outbound-codec-prefs to PCMU@20i,PCMA@20i so timing on inbound and outbound legs is the same
Info about async PTIME does not appear now in console, so I think now there are synchronous
But the problem is not solved - audio stream from local user is still lagging
Finally solved this problem by: 1. outbound-codec-prefs to PCMU@20i,PCMA@20i 2. setting inbound-codec-negotiation = scrooge Spent 4 hours on fixing that, maybe my solution will help someone with same problem in the future
Sounds like the linksys bug, do you have linksys or sipuras on the other end. The setting on the ptime there has a bug and you have to manually set it to 20. http://wiki.freeswitch.org/wiki/Interop_List#Linksys_Products Our ability to tolerate that might have an issue if it doesn't auto correct itself but then again why do we always have to marr our code to tolerate 10 year old bugs ;) On Thu, May 23, 2013 at 6:47 AM, Adam Raszynski <netcentrica@gmail.com>wrote:
Finally solved this problem by: 1. outbound-codec-prefs to PCMU@20i,PCMA@20i 2. setting inbound-codec-negotiation = scrooge
Spent 4 hours on fixing that, maybe my solution will help someone with same problem in the future
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Yes, it's related with Linksys PAP2T. The problem is that there are millions of this devices around the world :) I don't blame FreeSWITCH for that, changing codec params did the trick and now it works great as before.
participants (4)
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Adam Raszynski
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Alex Crow
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Anthony Minessale
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jay binks