[SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

[Paul Belanger]

On Tue, Aug 26, 2014 at 2:20 PM, Alex Villací­s Lasso
<a_villacis at 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?
>>
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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?
>
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

-- 
Paul Belanger | PolyBeacon, Inc.
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