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!
type=endpoint context=some_context disallow=all allow=speex allow=gsm allow=alaw allow=ulaw allow=speex16
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
type=auth auth_type=userpass password=[redacted]
type=aor remove_existing=yes max_contacts=1