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

[Olle E. Johansson]

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