[Asterisk-Users] Asterisk SIP Packet Time (20ms)

[Clif Jones]

Interesting.  For the record, the MultiTech MVP-130 comes with a default 
setting
of 60ms packets on all of its supported codecs.  I changed the packet 
sizes to
20ms because I had never heard of anyone using such large sample sizes.

Andres wrote:

>On Monday 22 December 2003 19:58, Rich Adamson wrote:
>  
>
>>>On Monday 22 December 2003 16:37, Andres wrote:
>>>      
>>>
>>>>On Monday 22 December 2003 15:36, Rich Adamson wrote:
>>>>        
>>>>
>>>>>>I have a question regarding the Asterisk Packet Time for SIP Calls.
>>>>>> It is hardcoded at 20ms but when I do an RTP Analysis on a stream
>>>>>>it is clear that these packets are not spaced out at 20ms.  In
>>>>>>general you see something like:
>>>>>>
>>>>>>Packet 50 - Delay 50ms
>>>>>>Packet 51 - Delay 5ms
>>>>>>Packet 52 - Delay 5ms
>>>>>>Packet 53 - Delay 50ms
>>>>>>Packet 54 - Delay 5ms
>>>>>>Packet 55 - Delay 5ms
>>>>>>
>>>>>>Is there anyway to space them out evenly at 20ms??
>>>>>>            
>>>>>>
>>>>>The 20 ms is not the inter-packet timing, its the relative content of
>>>>>what's within the packet. In other words, the packet contains 20ms of
>>>>>encoded voice.
>>>>>
>>>>>If the inter-packet times (delays) are large, as they would seem to
>>>>>be in your example, then something else is not right. Possibly a
>>>>>half-duplex ethernet connection, something else running on the
>>>>>server, router buffers, etc.
>>>>>
>>>>>On a typical * --> C7960 local call, I generally see from 1ms to 20ms
>>>>>inter-packet delays. Seldom (if ever) anything above 20ms.
>>>>>          
>>>>>
>>>>Thanks for your Input Rich.  I went ahead and tested this on our
>>>>production servers and sure enough the inter-packet times are 20ms. 
>>>>There must be something happening with our LAB Asterisk.  It could be
>>>>the CBQ traffic shaping software we have running on it.  I will fiddle
>>>>around with it to see if it changes anything.
>>>>
>>>>Thanks!
>>>>Andres
>>>>        
>>>>
>>>Ok...after some more testing, the traffic shaping software was not the
>>>culprit.  It turns out that if the UA is configured for 60ms of voice,
>>>then Asterisk will show this strange behaviour.  If we set the UA for
>>>20ms, then all works well.
>>>      
>>>
>>Cool!
>>
>>How did it get set to 60ms?
>>    
>>
>The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the 
>transmit packet size to 60ms (or multiple other values).  Asterisk will 
>receive 60ms and transmit 20ms times 3 packets, andit works quite well.  In 
>any case our SPA2000 problem was unrelated to the packet time.
>
>Regards,
>Andres 
>  
>
>>
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