[Asterisk-Users] GSM codec bandwidth

[Michael George]

On Fri, Sep 03, 2004 at 08:26:28AM +0200, steve at daviesfam.org wrote:
> On Thu, 2 Sep 2004, Michael George wrote:
> > I've a question about the bandwidth consumed by IAX2/GSM.
> > 
> > According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
> > for a voice encoding.
> > 
> > However, watching gkrellm when I initiate a call to Digium, it looks like the
> > channel is taking a consistent 5-6 kilo-bytes/sec.  That's a lot more
> > bandwidth than it should take.  Is there perhaps a setting I have wrong
> > somethere in the conf files?
> > 
> > I have:
> > bandwidth=low
> > disallow=all
> > allow=gsm
> > 
> > so it's surely using GSM and it should be gearing itself for a low-bandwidth
> > situation.
> 
> 
> The codec itself takes 13kbps, but by the time the codec frames are 
> wrapped in all the IP overhead it is a lot more.

Yes, I understand about overhead, but this is 4x the bandwidth usage.  Even if
that is 13kbps for each stream of audio (23kbps total), that is doubled by
(TCP/UDP)/IP overhead.  That struck me as a lot of overhead.  I guess, though,
that since the packets need to be sent quite frequently, that could happen.

If that is what others are experiencing, then I accept it.

> If you are sending several concurrent calls to the same place, you can 
> reduce the overhead by using trunking - which shares the IP overhead over 
> the concurrent calls.

That makes sense, and I've read that trunking pays off with even 2
conversations.

-- 
-M

There are 10 kinds of people in this world:
	Those who can count in binary and those who cannot.