Brad Templeton wrote:
On Sun, Jan 07, 2007 at 04:12:27PM +0000, Thomas Kenyon wrote:
Brad Templeton wrote:
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
Unless bandwidth between the * servers is a concern, then you're better
off keeping the link between servers as IAX. (preferably trunked)
The bandwidth of the audio stream dwarfs the bandwidth of signalling
traffic by orders of mangitude. So in fact, I think this is exactly
wrong. If bandwidth to or between the servers is a concern, that's
where you most want to not be in the audio path.
But if you have multiple RTP streams emnbedded in an IAX trunk, then the
IP overhead is significantly reduced.
AFAIK video should work for IAX2, there is explicit support for it.
(unlike h323).
Asterisk will only be able to pass the raw RTP traffic though, since it
doesn't have any video codecs (just format definitions).
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