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.
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?
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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?
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