Martin Langhoff wrote: > On Mon, Aug 11, 2008 at 4:35 AM, Tim Moody <[EMAIL PROTECTED]> 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/ _______________________________________________ Server-devel mailing list Server-devel@lists.laptop.org http://lists.laptop.org/listinfo/server-devel
