I would greatly appreciate if someone explains the following. Let’s say I have a DID which points to userID@fwd.pulver.com. I set an (inbound) trunk for userID@fwd.pulver.com and in my Inbound Routing I set “Callthru” destination for this trunk.
For out-bound calls, I use a provider “Gizmo Project” and have an (outbound) trunk for Gizmo Project. Thus when someone dials my DID number, he/she gets Gizmo’s dial tone and can then dial any PSTN.
Once a connection is established, how the audio stream is routed? Does it actually physically goes through your PBX server (and under what conditions)? If yes, is it recoded with actual G711/G729/etc codec implementations (and under what conditions)?
How the option “audio bypass” in trunk setup affects the above?
Thanks!
This post has been edited 1 time(s), it was last edited by ryb on 07.10.2006 at 17:09.
I do not think I can answer all f this however I can answer some:
1) there is no codec translation, therefore nothing gets recoded.
2) If trunk and extension are audio bypass YES , then audio stream will bypass server, assuming everything else allows reinvites ( I do not believe thet Gizmo AKA SIPPhone does). This audio bypass yes will also probably have issues if either endpoint is behind a firewall
3) g729 is not available on the PBX
4) you need not send the DID through FWD, you should be able to send it without the registration to FWD, but it can be unreliable!
Originally posted by supernettel
2) If trunk and extension are audio bypass YES , then audio stream will bypass server, assuming everything else allows reinvites ( I do not believe thet Gizmo AKA SIPPhone does). This audio bypass yes will also probably have issues if either endpoint is behind a firewall
Let's say, for a given connection (Callthru), we are having a good 2-way audio with audio bypass option set to "Yes" on both incoming and out-bound trunks. Does it necessarily imply that the audio stream is indeed bypassing the PBX server?
Zitat:
Originally posted by supernettel
3) g729 is not available on the PBX
Zitat:
Originally posted by i-p-tel
PBXes translates codecs if neccessary.
g729 included or excluded?
Just to clarify the terminology, my understanding is that "translation" = "audio stream gets re-coded from one codec format to another". Is that right?
To what extent it is important that as much of the audio traffic as possible will bypass the PBX server? What tweaking is/can be done to insure that? Which VoIP providers support that?
Thanks!
This post has been edited 1 time(s), it was last edited by ryb on 08.10.2006 at 19:33.
I suggest that you start sniffing packets or make diagnotic logs of the servers thatr you want to use
Note the sources and destinations of the media streams.
I was unaware of CODEC translation on the PBX, as it never has seemed to work for me , you can bet I will be testing.
"=" means they are the same thing. Audio Bypass is the same as CanReInvite. This will help toassure that the audio Is NOT passing through or being transcoded by the PBX, but does not guarantee it due to the capacity of the other servers to use it.
This post has been edited 1 time(s), it was last edited by sup on 08.10.2006 at 20:33.
RE: audio stream routing