asterisk users - Mar 2016 - Differences between Chan_SIP and PJSIP with NAT and STUN

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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Chirag Desai wrote:
> 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.
There should be nothing different, except for how you configure things. 
What is the full PJSIP configuration? What is the environment where 
Asterisk is running? Is ICE actually in use on the other side? What is 
the full SIP trace?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org