wuerhui at gmail.com
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 Posted: Mon Apr 07, 2014 11:09 am Post subject: [asterisk-users] asterisk-users Digest, Vol 117, Issue 7 |
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Hi Patrick,
Thanks a lot for your quick help. Yes, I configured the NAT options in
sip.conf.
BTW, I am using 12.1.1, will try 11.8.1 and see if I can make it work.
Thanks again,
William
=======================================
Date: Sat, 05 Apr 2014 23:38:32 +0200
From: Patrick Laimbock <patrick@laimbock.com>
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and SRTP
Message-ID: <534077D8.7000402@laimbock.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 04/05/2014 07:56 PM, William Wu wrote:
| Quote: | Hi experts,
I am trying Asterisk SRTP in my environment, and find that when
Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay
by Asterisk are local ports on the Asterisk server, media from the two
clients out of the NAT (for example from Internet) can not reach the
ports, and thus the two client can not establish the secure call via
Asterisk. I have set up a STUN server and configured in rtp.conf, but
seems Asterisk does not do STUN before it opens ports for SRTP. BTW,
Non-SRTP call can work though.
Anyone can give advice on how to make SRTP work in such an env?
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I have no problems with a TLS/SRTP call between a GSM with CSipSimple
and Asterisk 11.8.1 behind NAT. Have you configured the NAT options in
sip.conf?
externip=...
localnet=...
nat=...
You may also need to add/change the options below. Check the sip.conf
example file to see what these options do and use what's best for your
situation.
canreinvite=no
directmedia=no
directrtpsetup=no
HTH,
Patrick
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