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

[Rich Adamson]

I'm not sure under what circumstances (from an overall performance 
perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
suggest choosing the size is roughly equivalent to MTU size. The 60ms
setting should result in larger packets which might be okay for high
speed uncongested links and satellite links. However, the smaller 20ms
packets effectively allow "more opportunity" for others to talk on the
wire and would likely improve response time for all devices on the wire.

Rich
------------------------
> 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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> >>
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> >  
> >
> 
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