I'm running 3.1.4 on centos and I'm having some trouble with voice only going one way.
Both extensions will ring each other, but after they connect, voice will only travel in one direction. One extension hears fine, but can't talk.
I've totally opened up the firewalls (including port 5060) and I'm still having the trouble. I'm not behind NAT.
I've tried it using X-Lite, VoIP phones and ATAs in a number of combinations. The problem is across the board. X-Lite sometimes automatically disconnects after 32 seconds.
Does anyone have any suggestions on where to look? ANY help appreciated. This has been going on for weeks.
Hello,
the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is not send E2E but using the Kamailio. In this case you would have to redirect it.
In the content of INVITE and 200, 183 or ACK you should be able to see the IPs for RTP data. Compare these with your phones. O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX
Be sure, the RTP ports are open on your firewall, too. You can see the ports in the content, too. M=audio XXXXX RTP/AVP 8 101
Greetings Timo
-----Ursprüngliche Nachricht----- Von: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von CaptWho Gesendet: Mittwoch, 14. September 2011 05:01 An: users@lists.kamailio.org Betreff: [SR-Users] One way communications in Kamailio 3.1.4
I'm running 3.1.4 on centos and I'm having some trouble with voice only going one way.
Both extensions will ring each other, but after they connect, voice will only travel in one direction. One extension hears fine, but can't talk.
I've totally opened up the firewalls (including port 5060) and I'm still having the trouble. I'm not behind NAT.
I've tried it using X-Lite, VoIP phones and ATAs in a number of combinations. The problem is across the board. X-Lite sometimes automatically disconnects after 32 seconds.
Does anyone have any suggestions on where to look? ANY help appreciated. This has been going on for weeks.
-- View this message in context: http://old.nabble.com/One-way-communications-in-Kamailio-3.1.4-tp32460597p32 460597.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
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Thanks Timo, I'll look into that, just have to figure out what you just said. :confused: I'm at the bottom of the learning curve and I got dumped into this after the guy that was dealing with it disappeared.
Timo Klecker wrote:
Hello,
the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is not send E2E but using the Kamailio. In this case you would have to redirect it.
In the content of INVITE and 200, 183 or ACK you should be able to see the IPs for RTP data. Compare these with your phones. O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX
Be sure, the RTP ports are open on your firewall, too. You can see the ports in the content, too. M=audio XXXXX RTP/AVP 8 101
Greetings Timo
Hello,
On 9/14/11 6:13 PM, CaptWho wrote:
Thanks Timo, I'll look into that, just have to figure out what you just said. :confused: I'm at the bottom of the learning curve and I got dumped into this after the guy that was dealing with it disappeared.
it will help also to watch the sip signaling traffic, using ngrep or wireshark.
For example with ngrep, on the same server with kamailio:
ngrep -d any -qt -W byline port 5060
will capture all the incoming and outgoing sip messages. You can see if there are some retransmissions, where the messages are sent, thus being able to detect if there is anything wrong in routing. If you do nat traversal, then also sdp and some headers must be updated in the proxy server.
Cheers, Daniel
Timo Klecker wrote:
Hello,
the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is not send E2E but using the Kamailio. In this case you would have to redirect it.
In the content of INVITE and 200, 183 or ACK you should be able to see the IPs for RTP data. Compare these with your phones. O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX
Be sure, the RTP ports are open on your firewall, too. You can see the ports in the content, too. M=audio XXXXX RTP/AVP 8 101
Greetings Timo
-
CaptWho -
Daniel-Constantin Mierla -
Timo Klecker