On Tue, Aug 26, 2014 at 2:20 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
El 26/08/14 12:02, Alex Villacís Lasso escribió:
El 25/08/14 18:28, Alex Balashov escribió:
On 08/25/2014 07:25 PM, Alex Villacís Lasso wrote:
However, I do not find an equivalent to bridge mode in the rtpengine command-line parameters.
Bridging mode of this type is not supported by rtpengine.
If this is true, then mediaproxy-ng/rtpengine should not be announced in the Kamailio documentation (http://www.kamailio.org/docs/modules/4.1.x/modules/rtpproxy-ng.html) as a "drop-in" replacement. At the very least, this requires a documentation fix.
I'd agree 'drop-in' replacement is not correct. I ran into the same issues as you. Current there is no bridge-mode in rtpengine, I point you to an open issue about it [1].
How would somebody implement the following scenario using rtpproxy or mediaproxy-ng/rtpengine ?
- Server with 2 or more interfaces, at least one of which is public, and
at least one of which is private (LAN)
- Public interface runs webserver that publishes web phone (SIP.js or
similar) for websocket
- Webserver runs kamailio with access to both public and private
interfaces
- Websocket managed by kamailio, for SIP.js signaling
- Private interface gives access to LAN where at least one traditional SIP
client (UDP port 5060) registers with kamailio
- Phone call initiated through websocket should contact SIP client in
private LAN after proper authentication.
Can this be done at all with current technologies? How?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Just to clarify, the above scenario is not exactly what I want to implement. What I want to implement is interconnection between Kamailio (managing websockets) and Asterisk (which, by itself, could run websockets but is currently isolated).
The scenario is a multihomed server that runs asterisk in localhost only, and uses kamailio to simulate multiple domains, and to provide SIP presence support. Currently, rtpproxy works correctly (in traditional SIP) to bridge the RTP packets from localhost to each of the interfaces. The question is: can it be used to bridge DTLS-SRTP, without touching the (encrypted) payloads, and delegate the decryption to asterisk itself?
While my setup is not the same as yours, the fix would be to bind asterisk to the same interface as kamailio (different port for SIP), then simple allow rtpengine to relay to the single interface.
For the moment, I had to rework my firewall rule sets to allow asterisk (rtp) on the public interface.
[1] https://github.com/sipwise/rtpengine/issues/4