Podrška #18784: gsm codec, bandwidth usage na openvpn-u - voip - bring.out redmine

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Podrška #18784

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gsm codec, bandwidth usage na openvpn-u

Dodano od Ernad Husremović prije više od 15 godina. Izmjenjeno prije skoro 15 godina.

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04.11.2009

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Izmjenjeno od Ernad Husremović prije više od 15 godina

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.

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