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

[Andres]

On Tuesday 23 December 2003 10:59, Rich Adamson wrote:
> I'm not sure under what circumstances (from an overall performance
> perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
In our network we set UAs to use 60ms (using G729).  Actual data measurements 
indicate a call consumes about 13Kbps.  If we use 20ms then it consumes about 
25Kbps.  These are of course peer-peer calls since Asterisk itself does not 
support transmitting at 60ms.  We prefer 60ms due to the fact that some of 
our customers are using dial-up for their VoIP, and bigger delays are 
preferable to dopped packets.

Andres.

> 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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