asterisk users - Nov 2016 - SIP and RTP port and IP addresses

Hi all

I'd like to log the client IP addr and port used for SIP and RTP *during* in
a
call.

The IPs must be the real source IPs (internet accessible).

How are these parameters available from dialplan?

For instance, ${SIPURI} holds the internal "IP:port" if the client is
behind NAT.
I need the external IP:port

Regards

Ethy

http://www.voip-info.org/wiki/view/Asterisk+func+sip_header


On 2016-11-09 08:13 AM, Ethy H. Brito wrote:
> Hi all
>
> I'd like to log the client IP addr and port used for SIP and RTP
*during* in a
> call.
>
> The IPs must be the real source IPs (internet accessible).
>
> How are these parameters available from dialplan?
>
> For instance, ${SIPURI} holds the internal "IP:port" if the
client is behind NAT.
> I need the external IP:port
>
> Regards
>
> Ethy
>

Hi Ethy,


Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:
> How are these parameters available from dialplan?
> 
> For instance, ${SIPURI} holds the internal "IP:port" if the
client is behind NAT.
> I need the external IP:port

You can get the peer's signalling IP address from ${CHANNEL(recvip)} and the
RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}.
If you need more information (like the codecs used) you can find other channel
variables on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL

Please note that, if you have not disabled re-invites, the RTP address may
change while the call is running.



Max

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On Thu, 10 Nov 2016 00:35:54 +0100
Max Grobecker <max.grobecker at ml.grobecker.info> wrote:
> Hi Ethy,
Hi Max and All. 
> 
> 
> Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:
> 
> > How are these parameters available from dialplan?
> > 
> > For instance, ${SIPURI} holds the internal "IP:port" if the
client is
> > behind NAT. I need the external IP:port  
> 
> 
> You can get the peer's signalling IP address from ${CHANNEL(recvip)}
and the
> RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}. If you
need
> more information (like the codecs used) you can find other channel
variables
> on https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL
Hmmmm.

${CHANNEL(rtpsource)} is always returning something like
"0.0.0.0:ppppp" where
p=[0-9]

and ${CHANNEL(rtpdest)} returns the internal (not accessible) IP addr if the
caller is behind NAT, therefore, not what I need.

Wouldn't these two variables have correct values only after the callee
answers
the call??
> 
> Please note that, if you have not disabled re-invites, the RTP address may
> change while the call is running.
Interesting observation.

Thanx

Ethy