system toolbox » voip

http://lists.digium.com/pipermail/asterisk-users/2004-September/054167.html

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.

[Asterisk-Users] what is the best codec for low bandwidth? for quality?

... the GSM 06.10 is quite fair and works for everybody

http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2

http://www.voip-info.org/wiki/view/Bandwidth+consumption

http://blog.asteriskguide.com/bandcalc/bandcalc.php

openvpn nam donese 10-15 kbit/sec overhead !!

ali nama ionako treba taj vpn kanal pa kontam da nismo time nešto zijanili

711 (gizmo) + 9361 + 9162 = 210-220 kbit/sec

očitanja su sa tomato realtime graph-a

kvalitet zvuka u svim ovim testovima je skroz ok - (echo test)

problem nije u bandwitdh-u nego kada počnu da se javljaju peak-ovi i prekidi

tako da problem voip-a nije problem postojećeg adsl bandwidth-a nego stabilnost - latency veze i handliranje kad dođe do problema u konekciji