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

[Clif Jones]

Olle,

Here is an interesting site that goes into some of the troubleshooting 
techniques in Voip:
http://www.voiptroubleshooter.com/
Maybe it will help your FAQ!

Olle E. Johansson wrote:

> 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.
>>
>
> I gather from your reply that there are recommendations regarding the 
> ethernet connection
> on your Asterisk server? half-duplex seems bad.
> Could you elaborate a bit on that?
>
> /Olle
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>