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

> Joshua Colp wrote:
>
> 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?
>

The full configuration is here:

http://pastebin.com/XqZG1m5X

I am connection over TLS / SRTP on port 5063.
When I put in a stun server asterisk sends media to the phone's external IP.

The asterisk is has a public IP and internal IP. It is internet facing, and
is not behind NAT.

When I had ICE enabled on the snom, it didnt seem to make any difference.
PJ showed an ICE error.

The sip trace is here:

http://pastebin.com/fDxbk289

Thanks for your help.
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Chirag Desai wrote:
>
>     Joshua Colp wrote:
>
>     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?
>
>
>
> The full configuration is here:
>
> http://pastebin.com/XqZG1m5X
>
> I am connection over TLS / SRTP on port 5063.
> When I put in a stun server asterisk sends media to the phone's
external IP.
>
> The asterisk is has a public IP and internal IP. It is internet facing,
> and is not behind NAT.
Have you done a packet capture to see if the RTP from the remote device 
is hitting the machine to narrow things down?

-- 
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

> Joshua Colp wrote:
>>
>> Have you done a packet capture to see if the RTP from the remote device
>> is hitting the machine to narrow things down?
>>
>>
>>
Nope. When I run with RTP encryption on it seems that rewrite_contact does
not work in PJSIP.

When I turn off RTP some calls get media, some don't. If you look at the
SIP trace it seems like the rewrite_contact doesn't always take affect.
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Chirag Desai wrote:
>
>
>
>         Joshua Colp wrote:
>
>         Have you done a packet capture to see if the RTP from the remote
device
>         is hitting the machine to narrow things down?
>
>
>
> Nope. When I run with RTP encryption on it seems that rewrite_contact
> does not work in PJSIP.
>
> When I turn off RTP some calls get media, some don't. If you look at
the
> SIP trace it seems like the rewrite_contact doesn't always take affect.
The rewrite_contact shows as working fine in the SIP trace. The log 
shows the message as received over the socket, before modification. If 
it wasn't working then the BYE would be going to the internal IP 
address+port.

Nothing stands out in the signaling.

-- 
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

I'm dialling from the snom and every few calls asterisk sends media to the
phones external IP and it works!

And then now and again it sends the media to the phones internal IP and I
hear nothing. I'm really at a loss.
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