Hi,
For my next little project on my PBXes I would like to implement audio bypass for my FXOs and extensions (same room, same router, same lan).
FXOs are Sipura 3102, extns are Sipura 1000 etc, router is a Draytek 2820n on BT adsl (dynamic IP).
Sip ports are forwarded to each sip device (devices have fixed local addresses).
My first attempt resulted in lack of audio (or 1 way) between extensions.
Does anyone have any pointers or suggestions on how to proceed, or is my setup simply not suitable ?
Regards
For SIP Re-Invites to work, the SIP UAs ought to be on Public IPs. But in your case, since all the UAs are on the same LAN, you should try to port forward the ports used by the RTP stream.
Since all the internal SIP UAs seem to be SPAs, their default range RTP range spans from 16384 to 16482. Try to split or extend the RTP port range, so each SPA has a unique range, and subsequently port forward every range to each UA's private IP.
Also ensure each SIP registration (FXS or FXO port) has a unique SIP port number, and try not to use port 5060 on any of the SPAs, since the Draytek's FXOs use it.
Hi Diafora,
I tried forwarding unique rtp ports, no difference noted.
Sip ports are already unique and forwarded (5060 not used). Draytek isn't a voip model anyway.
Signalling is fine, just the audio is a problem
I was wondering about the Sipura 'nat support parameters', haven't changed those although I have seen totally contradictory advice in various forums ranging from 'all on' to 'all off' with every combination in between....
Ron
Before you start changing settings on your SPAs, try to enable the SIP ALG on your Draytek. Just SSH into it, and type "sys sip_alg ?" to get its' current state.
Hi Diafora,
Just checked and the sip_alg is off.
I remember I switched it off soon after purchase when investigating dropped calls and choppy audio. (Those issues were subsequently fixed by forwarding unique sip ports and setting the qos correctly).
I will have a play when I'm back from holiday...
Regards
Ron
Hi Diafora,
Setup a couple of extensions at home as a test environment - nothing made any difference, no audio in either direction. Eventually changed the CODECs from G726-32 to G711 and Voila worked first time extension to extension. I hadn't realised that requirement !!
Anyway will extend tests to trunks etc over the next few days,
Regards
Ron
Well, mixed results playing with trunks.
No luck at all with incoming Sipgate trunk, possibly doesn't support reinvites.
Managed one way (outgoing) audio with Voicetrading trunk by switching STUN and all NAT support parameters OFF in the ATA. RTP forwarding or using DMZ in domestic router made no difference (not much confidence in this old Azurewave router though).
This post has been edited 2 time(s), it was last edited by tel on 29.05.2009 at 01:17.
