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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