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Re: [Linphone-developers] Voice delay increasing during call


From: 宋智峰
Subject: Re: [Linphone-developers] Voice delay increasing during call
Date: Mon, 4 Jan 2016 15:36:43 +0800

Hi,
 
Anyone can share some idea?
 
Thanks,
Frank
2016-01-04

宋智峰

发件人:"宋智峰"<address@hidden>
发送时间:2015-12-14 17:19
主题:[Linphone-developers] Voice delay increasing during call
收件人:"linphone-developers"<address@hidden>
抄送:
 
Hi,
 
We have ported Linphone 3.7.0 to an ARM board with Freescale i.MX6 running on Yocot Linux. Everything looks fine, only that the voice delay is increasing during the call. For example, make call from normal IP phone to Linphone, after 2 hours, while talking on IP phone, the voice heard from Linphone has obvious delay (should >1s).
 
We are trying to find out the cause. One finding is, the clock skew is also increasing: at the call beginning the clock skew is -9.45ms, but 3 hours later, it becomes 3229ms and jitter buffer size is shown as 720ms (adatptive jitter buffer is turned off), as the linphone log shown below: 
 
[20151214-11:54:24.020] [message] bandwidth usage for call [0xfbef60]: audio=[d=80.5,u=81.0] video=[d=532.7,u=621.5] kbit/sec
[20151214-11:54:24.020] [message] Thread processing load: audio=17.863424 video=3.797420
[20151214-11:54:24.303] [message] sound/wall clock skew is average=-9.459320 ms
[20151214-11:54:25.009] [message] bandwidth usage for call [0xfbef60]: audio=[d=80.1,u=80.6] video=[d=521.0,u=611.6] kbit/sec
[20151214-11:54:25.009] [message] Thread processing load: audio=20.526340 video=4.368080
[20151214-11:54:25.281] [message] Sending RTCP SR compound message on session [0xfe6338].
[20151214-11:54:25.298] [message] audio_stream_iterate[0xfe7278]: local statistics available
 Local's current jitter buffer size:39.360001 ms
[20151214-11:54:25.383] [message] Sending RTCP SR compound message on session [0xfdf8a0].
[20151214-11:54:25.401] [message] video_stream_iterate[0xf7e6b0]: local statistics available
 Local's current jitter buffer size:69.177124 ms
 
... ...
 
[20151214-15:01:31.018] [message] bandwidth usage for call [0xfbef60]: audio=[d=80.8,u=81.3] video=[d=559.3,u=583.3] kbit/sec
[20151214-15:01:31.018] [message] Thread processing load: audio=17.859980 video=4.901045
[20151214-15:01:31.351] [message] Sending RTCP SR compound message on session [0xfe6338].
[20151214-15:01:31.366] [message] audio_stream_iterate[0xfe7278]: local statistics available
 Local's current jitter buffer size:720.000000 ms
[20151214-15:01:31.542] [message] sound/wall clock skew is average=3229.025896 ms
[20151214-15:01:32.002] [message] bandwidth usage for call [0xfbef60]: audio=[d=79.4,u=80.8] video=[d=556.4,u=579.4] kbit/sec
[20151214-15:01:32.002] [message] Thread processing load: audio=21.201956 video=5.428172
[20151214-15:01:33.006] [message] bandwidth usage for call [0xfbef60]: audio=[d=81.6,u=82.1] video=[d=564.9,u=590.2] kbit/sec
[20151214-15:01:33.006] [message] Thread processing load: audio=19.752945 video=5.418468
[20151214-15:01:33.543] [message] sound/wall clock skew is average=3229.011706 ms
 
Not sure the long clock skew is related to the voice delay issue, so any hints is appreciated.
 
Thanks and regards,
Frank
 
  
 
 
 
     
 
 
2015-12-14

宋智峰

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