Hi,

I was able to install Trixbox successfully on VMware and bought SPA400 to allow me to connect my PSTN lines to IP. I just have some few questions on codecs and IAX.

1.) Who controls the codec itself? I can see commercial IP phones and they said they support G729/G711. Does it mean that they are the one's defining the codecs needed? How does asterisk plays a role on controlling these codecs? Does it mean that the other end of the line should also be using the same codec?

2.) I have read from the forum that IAX2 is the way to go to consume bandwidth and is more NAT-friendly. Now my question would be, how does IAX2 and SIP differ when it comes to voice quality? Is it backward compatible with SIP? Or do you have to download new IP phone and softphone that supports IAX?

Hope someone could help me out. Thanks.

I don't know much about Trixbox, and I can't answer your second question, but I can definitely answer the first.

G729 and G711 are both patented codecs: they are "controlled" by the patent holders. The *ONLY* codec that is truly free is Speex, but since you didn't mention it I guess your phones don't support it.

Mike

Anyone can answer my second question? And please enlighten me on storkus answer, I want to know who controls the codec and is it dependent on the softphone/ip phone codec. Thanks.

sip and iax2 are unrelated protocols, but a sip device can talk to an iax2 device if
asterisk is in the middle. you won't have much of choice between sip and iax2 as far
as hardware phones go, since iax2 is not widely supported. but, you can find iax2 softphones.
iax2 is good for reducing bandwidth when multiplexing several conversations between
two asterisk servers into a single stream (read the wikipedia articles on iax2 and sip).
not sure how much bandwidth is saved with iax2 versus sip otherwise. voice quality is
affected by the choice of codec, which is a thing independent of iax2 or sip. asterisk
and a phone need to have at least one codec in common in order to be able to exchange voice.
in general, a phone using one codec can talk to a phone using another codec with asterisk
in the middle. an exception to that is g729, for which asterisk needs a transcoding
license; however no license is needed if asterisk is between two phones speaking g729.

Thanks for the info. So what asterisk does is to just convert the one codec to another so that the other end will be able to communicate effectively? But what if I've configured IAX2 extensions and would want an SIP IP Phone to call IAX2 extension, will this work? Can asterisk convert the signal from the other phone to the other end? Thanks.