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one way audio for the first 8 seconds

ndemou

By ndemou
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ndemou Newbie

ndemou

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We've found that when we call from yealink phones to other IP phones we frequently have one way audio for about 7 to 8 secs (specifically the caller doesn't hear the callee)

 

We did some tests and here are the results:

 

* we were calling from a yealink T28 to a Linksys SPA941 phone and vice versa

* the problem appeared in more than half of the calls (about 20 calls)

* setting pbxnsip general settings -> ports -> "lock codec during conversation" setting ON and OFF made no difference

* every one of the calls (both those without problem and those with the problem) resulted in the following line appearing twice in the pbxnsip log just when the calle picked up the phone:

"[7] ...... Different packet size (30 and 20), falling back to transcoding"

(or "Different packet size (20 and 30)" when calling the other way around)

 

Then we went to Linksys configuration -> advanced -> SIP and changed "RTP packet size" from 0.030 to 0.020 (apparently 20msec) and the problem disappeared along with the message in the logs. We did at least another 20 calls without any problem.

 

Here is what still puzzles us:

 

* the change of RTP packet size to 20msec in the Linksys phone solved the problem but will this bring up problems elsewhere? Maybe we should have changed the "RTP packet size" in the yealink configuration from 20 to 30msec because after all before introducing Yealink phones in our setup we had never seen this problem (and we had quite a long experience with many other phones). Then again we didn't find any such option in the web interface -- is there any?

* Is it that when pbxnsip has to do transcoding some times it does it the wrong way? We were making successive calls one after the other without any apparent difference between them so it seems to be failing randomly to us. If it's not a random failure within pbxnsip what is it?

 

UPDATE:

after making the above change in the production environment we still see calls with the "one-way audio for 8s" symptom. The don't produce any "falling back to transcoding" line in the logs and they still appear in calls between yealink and linksys phones. So the problem gets urgent again.

 

______________

PS

we have debug level 9 logs of both a failed and a successful call but since they are about 550Kbytes we thought we shouldn't post them unless asked.

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Vodia PBX Grand Master

Vodia PBX

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Is it possible to get the wireshark trace for this call?

 

Additionally, looking at the trace my first guess would be that the phone has problems with a full SDP answer (more than once codec). Can you quickly try to lock the codec or offer only one codec, e.g. G.711? We have seem other cases where the phone had a problem with G.729 where it just completely lost audio after some time. Maybe there are still some gremlins in the audio subsystem of the phone.

 

And do you have another phone for comparison? Maybe this is a network related issue.

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ndemou Newbie

ndemou

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[...] the phone [may have] problems with a full SDP answer

[...] Maybe there are still some gremlins in the audio subsystem of the phone.

And do you have another phone for comparison?

which of the two phones are you refering to? (the yealink or the linksys)

 

Can you quickly try to lock the codec or offer only one codec, e.g. G.711?

after reading through the forum we've tried both on and off options for the setting:

pbxnsip general settings -> ports -> "lock codec during conversation"

are yo refering to this setting? If yes as I've written above it made no difference

 

[...] or offer only one codec, e.g. G.711?

do you mean to go to pbx settings -> ports -> RTP: -> Codec Preference and leave only one codec available ?

 

And do you have another phone for comparison? Maybe this is a network related issue.

we have many phones available and we've only seen the problems when the one of the two phones is a yealink and the other is not

until now we've seen it in three networks -- each one with a different adsl modem/router

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Vodia PBX Grand Master

Vodia PBX

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I don't have wireshark on the pbxnsip server but I can run tcpdump following the instructions "Capturing with tcpdump for viewing with Wireshark"[1]

Will that be OK?

 

Yes, tcpdump is also fine. BTW from command line, you can also use tshark, that's the command-line version of Wireshark.

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  • 4 weeks later...
ndemou Newbie

ndemou

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[...] did you find any solution with this issue?

 

before we could reproduce the problem "in vitro" (it only happens randomly to us) we found out a hack that seems to have solved the problem in the sense that no user is complaining any more:

 

on the yealink phone navigate to phone -> voice -> JITTER BUFFER

set "Type" to "Fixed"

set all 3 delays to 120 (Min Delay,Max Delay,Normal Delay)

done

 

CAUTION: I don't understand the logic behind this hack and I haven't researched it. The only good thing is that I have a few phones with these settings for many days/weeks and nobody has complained.

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  • 2 weeks later...
HostCorp Newbie

HostCorp

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Thank you so much for the reply.

 

No user complaints is a fix in my books! - I was ready to ditch the yealinks and get snoms but have just applied this now and will watch it for the day and see how we go..

 

very excited that this may be the fix! I had looked at that jitter buffer setting and was tempted to set it to fixed but had not tried this to date.

 

Will let you know how we go.

 

Cheers,

Marcus.

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  • 4 months later...
mobileoffice Newbie

mobileoffice

Posted
Posted
before we could reproduce the problem "in vitro" (it only happens randomly to us) we found out a hack that seems to have solved the problem in the sense that no user is complaining any more:

 

on the yealink phone navigate to phone -> voice -> JITTER BUFFER

set "Type" to "Fixed"

set all 3 delays to 120 (Min Delay,Max Delay,Normal Delay)

done

 

CAUTION: I don't understand the logic behind this hack and I haven't researched it. The only good thing is that I have a few phones with these settings for many days/weeks and nobody has complained.

 

Can you tell me if the one way audio problem and fading in and out during the call is still fixed by this change. I have a client with the same problems and we have both G729 and G711a enabled. Was about to change all handsets as the client is NOT happy?

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  • 2 months later...
Vin Newbie

Vin

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Here's Yealink technical support:

 

Sorry that we didn't know about this report here until now.

This is a random issue may happen in some special environment. We are not sure whether it makes sense but we tried to fix it on our point. There are V51 beta firmwares you can download to have a try:

T28 V51 Beta http://www.yealink.com/fae/2.51.0.6.rar

T26 V51 Beta http://www.yealink.com/fae/6.51.0.6.rar

T22 V51 Beta http://www.yealink.com/fae/7.51.0.6.rar

T20 V51 Beta http://www.yealink.com/fae/9.51.0.6.rar

 

V51 change log http://www.yealink.com/fae/Yealink SIP IP Phones V51 Firmware Change Log.rar

However, this bug fix is not declared in the changelog.

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