[Asterisk-Users] PAP-2 Conferencing Problems

[Stuart Elvish - Dallas Delta Corporation Pty Ltd]

Dear Andres,

Thank you for your suggestion. Someone else here in Melbourne has 
suggested that I also check the RTP timing in the meetme.conf file which 
has been commented out (just the default configuration). From there I 
found that you are right in your suggestion of the 0.020 instead of the 
0.030. The meetme.conf RTP buffer is in multiples of 20 milliseconds and 
therefore I assume can't handle RTP sizes of 30 milliseconds.

Thank you so much for your help on this - it has made all my stress 
disappear (until Monday when I will have more problems...) and initial 
testing seems to prove that it works.

Kind Regards
Stuart Elvish


Andres wrote:
> Stuart Elvish - Dallas Delta Corporation Pty Ltd wrote:
>
>> Just come across a problem - we have sent out heaps of PAP-2 ATA's 
>> and just discovered that when joined in a conference they are choppy 
>> on the up leg (so the other users in the conference will hear them 
>> with a choppy sound) but the down leg is perfectly fine (so the end 
>> user can hear the conference participants perfectly).
>>
> Is that into a meetme conference?  If so I have noticed that you have 
> to change the default RTP Size (on the PAP2 or Sipura) to .20 instead 
> of .30.  Don't know why that is but would be very interested in 
> knowing if it fixes your issue.  I have tried also with .60 and it is 
> not only choppy, its totally unintelligible.
>
>> I have tested the same setup with different brands of ATA's and with 
>> IP phones and there aren't any problems. I have also tested a couple 
>> of different codecs (g729 and ulaw) and the problem seems to still 
>> exist.
>>
>> The problem happens when the ATA is  both internal and external to 
>> the VoIP server network.
>>
>> Does anyone have any suggestions?
>>
>>
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>>
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>

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