[asterisk-dev] s?rtp via SIP/TLS/TCP

[James Cloos]

I'm not sure whether this is a bug, so I'm starting here.

My remote asterisk (debian's compile of 13, currently 13.1.0) and my
snom had been unable to rtp for some time.  I still use chan_sip.

It took a few hours of testing, but I determined that when the phone
registers and/or invites over tls, asterisk refuses to do either un-
encrypted rtp or srtp unless force_avp=yes.

The symptom was a hangup w/ unknown cause as soon as it was time to
start rtp.  Ie, right after the OK.  Even with verbose and debug set
to 9 no explanation was logged.  When I tried adding force_avp=yes
everything started working again.

I don't know when it stopped working (I don't get enough calls and
send outgoing via a different asterisk); I presume that it stopped
working when force_avp was added, but cannot confirm that.

Is force_avp supposed to be required for non-dtls rtp when sip is
secure?

-JimC
-- 
James Cloos <cloos at jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6