[Freeswitch-users] IAX Trunking
Daniel Swarbrick
freeswitch at pressure.net.nz
Wed Jun 18 03:17:42 PDT 2008
You would probably save more bandwidth by switching to a lower bandwidth
codec. IAX trunking only reduces bandwidth wasted by IP headers of
individual RTP streams, by combining multiple small RTP packets (from
different RTP streams) into larger packets. It does not do anything
magical to the RTP data itself.
This is only useful if all those RTP streams are going to the same
destination of course.
With SIP, you could try increasing the packetization delay from the
usual 20 or 30ms up to something like 120ms. This means more RTP data
gets packed into a single packet, and hence fewer of them are required
in any given RTP stream. It does have side effects however - if your
link is not reliable enough, and you lose the occasional packet, that's
120 ms of audio you just lost - probably quite noticeable. Depending on
the jitter buffer being used, it may also want at least a few packets in
the buffer before it starts playing audio. If 3 x 120ms packets is what
it wants, there will be a 360ms delay at the start of the call before
the first audio is heard.
On Wed, 2008-06-18 at 11:36 +0200, David Villasmil wrote:
> Thanks for the reply,
>
> I was just curious because we have many gws doing IAX trunking to
> save bandwidth as the gws are behind ADSL.
>
> Thank you again.
>
> d
>
> On Wed, Jun 18, 2008 at 11:18 AM, David Knell <dave at 3c.co.uk> wrote:
> Hi David,
>
> I believe the answer to be 'no' - the workaround we use is to
> run * and FS on the same
> box, and have * act as an IAX<->SIP gateway. Works fine, but
> probably restricts capacity
> in the limit.
>
> Cheers --
>
> Dave
> > Can FS do IAX Trunking?
> >
> > David
> >
> >
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>
> --
> David Knell, Director, 3C Limited
> T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031
> http://www.3c.co.uk
>
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