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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.<br><br>All the UACs are behind NAT, and there are two kinds based on nat_uac_test (in our case set to 18):<br><br>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.<br><br>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.<br><br><br>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. <br><br>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).<br><br>Has anyone experienced such intermittent one-way audio issues? Can the router cause this somehow (which is configured by our provider)?<br><br>Thanks a lot,<br>Matt<br> </div></body>
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