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 <[email protected]> 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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