[asterisk-users] One-way audio with media_address

Richard Kenner kenner at gnat.com
Tue Sep 4 07:34:54 CDT 2012

I'm migrating from Asterisk 1.6.2 to 10.7.0.  In 1.6.2, I made a small
patch to allow specifying an address for RTP media.  That worked.  In
10.7.0, this appears to be built in with "media_address", but it doesn't
work for me.

My Asterisk server has multiple addresses, all global address on two
different /24's with different routing policies via BGP.  I'm connecting to
a phone that's over NAT.  I have "nat=yes" in the "general" section of
sip.conf.  Everything works fine with the default.

But if I specify media_address to be the Asterisk server's address on the
other /24, I get one-way audio.  I can see with "sip debug" that the proper
address is being given in the SDP data.  Audio from the phone is fine.
Audio *to* the phone starts out with maybe 1-2 seconds of very garbled
audio, then goes quiet.

Running traceroute shows that data comes from the phone *to* Asterisk on
the desired /24, but goes out with a source address from the other /24 (the
default address).  I'm not sure if this is the problem or not, but in any
event, I think the source address for RTP should be the one in
"media_address" and want it that way for my purposes anyway.  Is there a
way to configure this to happen?  If not, where should I look to make a
patch?  And is this likely the reason for the one-way audio or is something
else the likely cause?

More information about the asterisk-users mailing list