[asterisk-users] PJSIP TLS sometimes RTP, sometimes no RTP

[George Joseph]

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
"nightly" version?

On Tue, Jan 19, 2016 at 1:20 PM, Chirag Desai <djchillerz at gmail.com> wrote:

> Hi,
>
> 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!
>
> Thanks!
>
> [someuser]
> type=endpoint
> context=some_context
> disallow=all
> allow=speex
> allow=gsm
> allow=alaw
> allow=ulaw
> allow=speex16
> allow=speex32
> allow=g722
> auth=someuser
> aors=someuser
> direct_media=no
> media_encryption=sdes
> media_encryption_optimistic=yes
> rtp_symmetric=yes
> force_rport=yes
> rewrite_contact=yes
> ice_support=yes
>
> [someuser]
> type=auth
> auth_type=userpass
> password=[redacted]
> username=someuser
>
> [someuser]
> type=aor
> remove_existing=yes
> max_contacts=1
>
> Thanks
>
> C
>
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