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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