[SR-Users] Scaling websockets over SIP
jitterbuffer at gmail.com
Wed Mar 1 07:41:16 CET 2017
mistakenly pressed sent earlier, here is complete query.
If we want to use multiple kamailio servers for WebRTC based user agents,
what would be the best practice:
1) WS/WSS Outbound/Edge Proxy
In this case would it not become a single point of failure? e.g. if
we have a wss edge proxy, would registrars under it be on plain SIP? if so
this WSS Edge Proxy should handle media then ? or the nodes should handle ?
2) Each UA mapped to a specific SIP URI and then routing of calls based on
shared DB socket id (to the registrar server of that UA) - it is not a
generic approach as each proxy will have its own address, making client
side to be restricted to use one proxy only.
Looking forward to some directions here. Thanks
On Wed, Mar 1, 2017 at 11:25 AM, Jade SZ <jitterbuffer at gmail.com> wrote:
> Hi Guys,
> I am using kamailio/rtpengine and webrtc over sip client for a/v calls and
> all works fine.
> Now I want to scale it to 2 or more than 2 kamailio servers. As per search
> on mailing list I could:
> 1) Websocket Outbound/Edge Proxy
> 2) Each UA client to register with its
> Jade SZ
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