[asterisk-users] PJSIP TLS sometimes RTP, sometimes no RTP
george.joseph at fairview5.com
Tue Jan 19 17:05:42 CST 2016
With the exception of media_encryption_optimistic=yes and ice_support =
no, my setup looks like yours and I'm not having any problems with
CSipSimple, even with SRTP mode = mandatory. I assume your server has a
public IP address and there's no NAT involved on the server side? Oddly
enough, I have ICE and Aggressive ICE turned on in CSipSimple.
CSipSimple in the Play store is a little stale. Have you tried the
On Tue, Jan 19, 2016 at 1:20 PM, Chirag Desai <djchillerz at gmail.com> wrote:
> I have a PJSIP account configured as below. I am testing with the Echo
> Test application on Asterisk 13 and using CSipSimple.
> I can create a call with TLS and SRTP, however for some reason only 1 in
> every 5 calls has audio.
> When I connect over WiFi, I have audio every single time. When I connect
> over 3G/4G I only get audio every now and then.
> Sometimes pjsip shows: Probation passed - setting RTP source address to
> [public ip:port] and I get audio when using a mobile network.
> Most of the time though asterisk shows it's playing the demo echotest
> file, but there doesn't appear to be any RTP and I hear no audio.
> I'm using TLS and SRTP (SDES) Mandatory. I've tried various codecs too.
> I've tried STUN and ICE but with little luck.
> Ideas would be greatly appreciated!
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