[Server-devel] VoIP

[Sameer Verma]

Martin Langhoff wrote:
> On Mon, Aug 11, 2008 at 4:35 AM, Tim Moody <timmoody at sympatico.ca> wrote:
>   
>> What are the bandwidth requirements for these various voip strategies, sip,
>> iax2?
>>     
>
> Not sure (google away!) - but the latency requirements very tight for
> many (most?) of our deployments.
>
> cheers,
>
>
>
> m
>   

Hi,

Bandwidth requirements will depend on the underlying codec used to 
compress the audio. If you look at 
http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2 
you'll see approx. bandwidth numbers + overhead for different codecs. 
Speex info is at http://www.voip-info.org/wiki/view/Speex

SIP simply establishes the connection and then hands it over to RTP 
(basically, SIP is establishing RTP ports on both sides). The 
significant difference between IAX and SIP is that IAX will use the same 
port for establishing the connection *and* for carrying the signal 
across (UDP 4569), which makes it easier to use across NATed networks. 
I've used IAX over three NATs (just for fun) and it still works:-)

SIP over NAT is troublesome, the problem being that SIP establishes the 
RTP port of the *private* IP (behind the NAT fw), which isn't routable 
from the public side...it will work, but requires port forwarding or 
tunneling. See http://freshmeat.net/articles/view/2079/ for more details.

So, we will need to pick a transport mechanism (SIP, IAX2, etc) and a 
codec that is good enough for low bandwidth requirements. Then there is 
the issue of jitter (jitter...sounds...like...this) which is now handled 
satisfactorily in Asterisk for IAX-based systems.

Sameer

-- 
Dr. Sameer Verma, Ph.D.
Associate Professor of Information Systems
San Francisco State University
San Francisco CA 94132 USA
http://verma.sfsu.edu/
http://opensource.sfsu.edu/