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 <[email protected]> To: [email protected] Subject: Re: [asterisk-users] Asterisk and SRTP Message-ID: <[email protected]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 04/05/2014 07:56 PM, William Wu wrote: >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? 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 > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
