Discussion:
[asterisk-users] No RTP from asterisk?
Peter Serwe
2010-02-27 20:42:56 UTC
Permalink
I've got an asterisk installation of 1.4.30-RC2 running, and while I can
register lines and get call setup to pass, for some reason no RTP is being
generated or received by asterisk.

Debug doesn't seem to give me too much of relevance about it, especially rtp
debug.

I had a few other small issues, like trying to negotiate G729 when it's not
capable, but since then, I've changed everything back to G711.

I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as
to how to troubleshoot this further.

Can anyone point me in the right direction?

Peter
--
Peter Serwe
http://truthlightway.blogspot.com/
Tri Tu
2010-02-28 06:05:26 UTC
Permalink
RTP is only firewall issue. Make sure that you can pass traffic from your client to the asterisk server. If it's on the same LAN, there shouldn't be any issue with RTP unless the Asterisk is setup with firewall to block RTP traffic (default is from 10000 - 20000 upd)

Asterisk doesn't support G29 (pass-through is OK) but if you want to connect from your client to asterisk server with G729, you need to buy license. Using G711 is free and it taking about 68kbp.





________________________________
From: Peter Serwe <***@gmail.com>
To: asterisk-***@lists.digium.com
Sent: Sat, February 27, 2010 12:42:56 PM
Subject: [asterisk-users] No RTP from asterisk?

I've got an asterisk installation of 1.4.30-RC2 running, and while I can register lines and get call setup to pass, for some reason no RTP is being generated or received by asterisk.

Debug doesn't seem to give me too much of relevance about it, especially rtp debug.

I had a few other small issues, like trying to negotiate G729 when it's not capable, but since then, I've changed everything back to G711.

I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as to how to troubleshoot this further.

Can anyone point me in the right direction?

Peter
--
Peter Serwe
http://truthlightway.blogspot.com/
Peter Serwe
2010-03-01 01:41:38 UTC
Permalink
I checked the firewall, iptables -L showed no rules whatsoever. No other
traffic has indicated it was blocked, iptables was set in allow all
everywhere mode.

I went ahead and turned it off, still don't have RTP. No audio either
direction via lines registered.

G729 is completely disabled from all trunk groups and users, only using G711
at this point.

Peter
Post by Tri Tu
RTP is only firewall issue. Make sure that you can pass traffic from your
client to the asterisk server. If it's on the same LAN, there shouldn't be
any issue with RTP unless the Asterisk is setup with firewall to block RTP
traffic (default is from 10000 - 20000 upd)
Asterisk doesn't support G29 (pass-through is OK) but if you want to
connect from your client to asterisk server with G729, you need to buy
license. Using G711 is free and it taking about 68kbp.
------------------------------
*Sent:* Sat, February 27, 2010 12:42:56 PM
*Subject:* [asterisk-users] No RTP from asterisk?
I've got an asterisk installation of 1.4.30-RC2 running, and while I can
register lines and get call setup to pass, for some reason no RTP is being
generated or received by asterisk.
Debug doesn't seem to give me too much of relevance about it, especially rtp debug.
I had a few other small issues, like trying to negotiate G729 when it's not
capable, but since then, I've changed everything back to G711.
I have connected to it, a SIP trunk, 3 registered users and I'm at a loss
as to how to troubleshoot this further.
Can anyone point me in the right direction?
Peter
--
Peter Serwe
http://truthlightway.blogspot.com/
--
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asterisk-users mailing list
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Peter Serwe
http://truthlightway.blogspot.com/
Duncan Turnbull
2010-03-01 02:21:23 UTC
Permalink
I checked the firewall, iptables -L showed no rules whatsoever. No other traffic has indicated it was blocked, iptables was set in allow all everywhere mode.
I went ahead and turned it off, still don't have RTP. No audio either direction via lines registered.
G729 is completely disabled from all trunk groups and users, only using G711 at this point.
Peter
The asterisk rtp debug should show if asterisk is sending audio or receiving packets but its not nearly as useful as tcpdump.

tcpdump udp port 5060 -s0 -A will give you all the SIP.

But just dumping all traffic between asterisk and the host will give you a view on RTP - you should see it take off when a call is setup if its not blocked

You should see a SIP Invite to setup a call with the audio destination - this should be your asterisk box and the far end depending on who is doing what. You should look at the address and also whether both sides are providing a mutually acceptable audio formats. If there are no agreed audio formats you won't get rtp. The c= in the session setup indicates the addresses each site is using for media.

Cheers Duncan
--[ UxBoD ]--
2010-03-01 18:56:41 UTC
Permalink
----- "Tri Tu" <***@yahoo.com> wrote:



RTP is only firewall issue. Make sure that you can pass traffic from your client to the asterisk server. If it's on the same LAN, there shouldn't be any issue with RTP unless the Asterisk is setup with firewall to block RTP traffic (default is from 10000 - 20000 upd)

Asterisk doesn't support G29 (pass-through is OK) but if you want to connect from your client to asterisk server with G729, you need to buy license. Using G711 is free and it taking about 68kbp.






From: Peter Serwe <***@gmail.com>
To: asterisk-***@lists.digium.com
Sent: Sat, February 27, 2010 12:42:56 PM
Subject: [asterisk-users] No RTP from asterisk?

I've got an asterisk installation of 1.4.30-RC2 running, and while I can register lines and get call setup to pass, for some reason no RTP is being generated or received by asterisk.


Debug doesn't seem to give me too much of relevance about it, especially rtp debug.


I had a few other small issues, like trying to negotiate G729 when it's not capable, but since then, I've changed everything back to G711.


I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as to how to troubleshoot this further.


Can anyone point me in the right direction?

Peter
--
Thanks, Phil

http://issues.asterisk.org/view.php?id=16929
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