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

[Chirag Desai]

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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