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https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+%28ICE%29+in+Asterisk
Feb 04, 2014 · The Interactive Connectivity Establishment protocol, or ICE, is a relatively recent and promising approach to resolving these kinds of problems. Support for ICE was added to Asterisk in version 11. ICE is a standardized mechanism for establishing communication suitable for live media streams between software agents running behind NAT firewalls.
https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support
Apr 30, 2012 · If remote SDP is parsed and ICE candidates present these are passed into res_rtp_asterisk which stores them until ICE negotiation should occur. After all ICE candidates are parsed negotiation and connectivity checks begin. ICE support uses the ICE session API and has been added into the set of operations for sending and receiving packets.
https://www.didforsale.com/disable-icesupport-asterisk
This is for customer using Asterisk flavored PBX. icesupport is enabled by default that causes the call to hangup right after 200 OK. As of writing of this document, DIDForSale registered sip trunk with asterisk does not support icesupport. Ice support can be disabled in /etc/sip.conf.
http://forums.asterisk.org/viewtopic.php?t=94147
Jul 21, 2015 · re: asterisk ice support by navaismo » Tue Jul 21, 2015 9:53 am If I recall some debian users uses another repo for the uuid libraries in order to get the support in asterisk.
https://community.freepbx.org/t/freepbx-13-ice-support-button-missing/42479
Good day Kindly note that we have a FreePBX 13 setup running Asterisk 13 on Centos 7. We had WebRTC extensions registered on the PBX and all was working well until we noticed that one of the phones was not able to receive incoming calls. We noticed then that if you go to the extension settings, the ICE support button was no longer available. We can work around the issue by going to /etc ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
Sep 22, 2016 · When building Asterisk 11, to get ICE support you'll need the UUID development library (uuid-dev for Debian, libuuid-devel for CentOS) library. If you don't have ICE support, then you'll likely run into audio issues in several scenarios, specifically when attempting to traverse NAT, as WebRTC uses ICE,STUN,TURN to do this.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip
Jul 05, 2015 · ice_support. Boolean. no. false. Enable the ICE mechanism to help traverse NAT . identify_by. Custom. username. false. Way(s) for Endpoint to be identified . redirect_method. Custom. user. false. How redirects received from an endpoint are handled . mailboxes. String false. NOTIFY the endpoint when state changes for any of the specified ...
https://blogs.asterisk.org/2017/08/23/webrtc-asterisk-14/
Our implementation of this has improved since the beginning to properly support secure WebSockets and also SIP over secure WebSockets. If you are wanting to get started in WebRTC with Asterisk this is the easiest option to use, with client libraries for the web browser being easily available. ICE. ICE in WebRTC is used for NAT traversal.
http://forums.asterisk.org/viewtopic.php?p=202031
Aug 13, 2014 · Hello, First, sorry for any mistakes in English. I'm using a translator to help me. I know there are hundreds of topics around the internet with people struggling with WebRTC and the asterisk and great guides help (as done by the user navaismo) , who can guide us a lot, but I'm a few days trying to get the work in WebRTC my platform and unsuccessful.
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