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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