[OpenSIPS-Users] intermittent one-way audio

[Matt Hamilton]

Sammy, thanks a lot for your suggestions.


>Can you track the call with one-way audio. That could be a big deal if there are alot of servers in your asterisk farm. If you can trace which asterisk server it happens >and verify its sip settings "externip" - also "rtp set debug on" for a brief time interval can tell you the story. 

It happens on all of them, but then they are all configured the same way. All servers have public IPs, and both outboundproxy and bindaddr are set to the public ip. externip is not set (commented out), but Asterisk sets SDP (c) field correctly, and SDPs seem to be negotiated well. As I said, this happens only for phones behind Cisco routers and it happens intermittently - they work ok for 10-15 minutes, then we see the problem. I can't see any abnormal/incorrect SDP data while the problem is happening. I haven't done "rtp set debug on" yet (I'll do that next), but wireshark shows the RTP packets are sent to the correct IP (how the Cisco router handles them might be causing the problem of course).

I will ask our provider to see if there are any ACL restrictions on the router and especially if there is anything in between that might cause a timeout as you suggested. 

> Do you see any re-INVITES exchanging between the phones and maybe both 
the A & B parties trying to directly communicate with each other 
!!? 

All the re-INVITES seem to be going thru Opensips and Asterisk, but I will try to capture the traffic between the phone and the router in our branch to see if there are anything going on between UACs.

Regards,
Matt



>Do you've any CISCO PIX/ASAthingie in your network ? Maybe it times-out/expires the connection stream due to inactivity !?>These are few things I could think of troubleshooting such an issue. Maybe some other hints or details from you can help focus in one particular area.

>Regards,>Sammy


On Thu, Mar 29, 2012 at 5:23 AM, Matt Hamilton <mistral9999 at hotmail.com> wrote:





We are using Opensips as a load balancer/dispatcher for Asterisk servers. All these servers are in a DMZ and have public IPs. SIP traffic goes thru Opensips, but RTP is between Asterisk servers and UACs.

All the UACs are behind NAT, and there are two kinds based on nat_uac_test (in our case set to 18):


1. The ones for which flag 2 (the "received" test) applies (address in Via is compared against source IP address of signaling). These are mostly behind firewalls, and source and via ports are the same - 5060.


2. The ones for which flag 16 applies (if the source port is different from the port in Via). These phones are directly connected to a Cisco router thru a switch.


We are having intermittent one-way audio problems for the clients in 2 in an environment where a client puts a call on hold and the other one picks up. The phones work properly without audio issues for 10-15 minutes, then one way-audio happens. We can't find anything out of the ordinary in the SDP fields; all the IPs seem to be correct. 


BTW, phones in 1 above work fine (all the time), and all the phones are exactly the same (for both 1 and 2 - same brand, firmware, configuration).

Has anyone experienced such intermittent one-way audio issues? Can the router cause this somehow (which is configured by our provider)?


Thanks a lot,
Matt
 		 	   		  

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