1. Brekeke Product Name and version:PBX 2.0
2. Java version:
3. OS type and the version: Windows XP SP2
4. UA (phone), gateway or other hardware/software involved:
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : LAN1
6. Your problem: Use of other codec's
Hi, I want to use wideband codec's like Speex and G722 and want to disable RTP relay completely. I have tried with both the settings in SIP dialplan and User/ARS/Options in PBX settings, but for some reasons the only codec's that are accepted by SIP server/PBX are PCM.
Thanks in advance for any tip/suggestions
Turn off RTP relay completely
Moderator: Brekeke Support Team
-
voipwell.com
- Posts: 528
- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
Hi,
You should be able to turn off rtp relay in the "from pbx" dial plan using these in the deploy. If you are not using pbx then just make sure these entries are in the dial plan that's active when making the calls out.
&net.rtp.relay.localhost=off
$rtp=false
We do it all the time for faxing and xlite video. It will allow both the video and t.38 codecs when doing it this way so I'm sure it will allow your wide band codecs.
Good luck!
Nick
You should be able to turn off rtp relay in the "from pbx" dial plan using these in the deploy. If you are not using pbx then just make sure these entries are in the dial plan that's active when making the calls out.
&net.rtp.relay.localhost=off
$rtp=false
We do it all the time for faxing and xlite video. It will allow both the video and t.38 codecs when doing it this way so I'm sure it will allow your wide band codecs.
Good luck!
Nick
Hi Nick, thanks for the info but just to understand you correctly, should this be added to the SIP server dialplan or in the PBX ARS settings?
In the SIP server dialplan I have the three default "from PBX" entries: PBX Prefix, From PBX1 and From PBX2 should I add this do these ones or define a new "From PBX" higher priority?
I did try to define a new entry in the dialplan but the &net.rtp.relay.localhost=off was colored in red in the dialplan, so I think this is an indication of that it's not valid.
BR/Olle
In the SIP server dialplan I have the three default "from PBX" entries: PBX Prefix, From PBX1 and From PBX2 should I add this do these ones or define a new "From PBX" higher priority?
I did try to define a new entry in the dialplan but the &net.rtp.relay.localhost=off was colored in red in the dialplan, so I think this is an indication of that it's not valid.
BR/Olle
-
voipwell.com
- Posts: 528
- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
Hi,
Remember that the call has two parts. For outgoing calls you have the first leg which is the ip phone calling the pbx (to pbx) and the second leg which is the pbx calling your isp(from pbx). Go ahead and make a call and look at the active sessions in the sip server. Click on the detail and note both the dial plans being called. Then go to the sip server and add the above entries to both dial plans and it should work. Do the same thing for an incomming call to be sure you get it right.
Another consideration is that if you turn off all rtp relay then Brekeke's session border controller that handles NAT traversal will not work and you could end up with no audio going to ip phones that are not on your local brekeke sip server network lan or that don't have port forwarding on their router.
Don't worry about the dial plan entries that turn red. Just make sure you typed it right and ignore the red color. It is a common thing for many dial plan entries.
Nick.
Remember that the call has two parts. For outgoing calls you have the first leg which is the ip phone calling the pbx (to pbx) and the second leg which is the pbx calling your isp(from pbx). Go ahead and make a call and look at the active sessions in the sip server. Click on the detail and note both the dial plans being called. Then go to the sip server and add the above entries to both dial plans and it should work. Do the same thing for an incomming call to be sure you get it right.
Another consideration is that if you turn off all rtp relay then Brekeke's session border controller that handles NAT traversal will not work and you could end up with no audio going to ip phones that are not on your local brekeke sip server network lan or that don't have port forwarding on their router.
Don't worry about the dial plan entries that turn red. Just make sure you typed it right and ignore the red color. It is a common thing for many dial plan entries.
Nick.
Hi Nick, still not working. I checked the session parameters and added the enties and it made no difference and att the bootom it says for both legs that rtp relay is off.
This is for internal calls so there is actually no ISP involved and all is behind NAT
I have the server on 192.168.1.12 and two clients on 192.168.1.100 & 101.
I have sip users on 300@192.168.1.12 and 301@192.168.1.12 and pbx users 300 and 301.
When I make a direct call from one client to the other and only uses the SIP server as a proxy all codec's are offered but when I make a call from 300 to 301 through the SIP server (sip:301@192.168.1.12) the only codec offered on the second leg (PBX to 301) is PCM.
I'm not sure if this explains by problem in more detail?
Thanks for your help / Olle
This is for internal calls so there is actually no ISP involved and all is behind NAT
I have the server on 192.168.1.12 and two clients on 192.168.1.100 & 101.
I have sip users on 300@192.168.1.12 and 301@192.168.1.12 and pbx users 300 and 301.
When I make a direct call from one client to the other and only uses the SIP server as a proxy all codec's are offered but when I make a call from 300 to 301 through the SIP server (sip:301@192.168.1.12) the only codec offered on the second leg (PBX to 301) is PCM.
I'm not sure if this explains by problem in more detail?
Thanks for your help / Olle