I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip. In my snom 760 the setup for these two accounts is identical.
When I call echo test from the account using chan_sip audio comes through fine. When I call echo test from the account using pjsip there is no audio. With rtp set debug on, I can see that audio is being sent to the snom's internal IP 192.168.0.x I can add a stun server in the config for this account and RTP flows to the Public IP and I get audio. I was wondering why there is a difference between pjsip and chan_sip so that one works without stun and the other requires it. Does anybody know why? Maybe my settings are off in pjsip. Here's how I have my endpoint configured: [test] type=endpoint context=dial_out disallow=all allow=alaw allow=speex allow=speex16 allow=speex32 allow=gsm allow=ulaw allow=g722 auth=test aors=test direct_media=no media_encryption=sdes media_encryption_optimistic=yes rtp_symmetric=yes force_rport=no rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ice_support=yes ;This is specific to clients that support NAT traversal ;for media via ICE,STUN,TURN. See the wiki at: ;https://wiki.asterisk.org/wiki/x/D4FHAQ ;for a deeper explanation of this topic. [test] type=auth auth_type=userpass password=redacted username=test [test] type=aor remove_existing=yes max_contacts=2 qualify_frequency=60 Looking forward to your thoughts. Kind Regards, C
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