[VoiceOps] Ideas for Building Inbound Redundancy
nick at flhsi.com
Tue Feb 7 13:58:29 EST 2017
Markus, Thanks for the info!
>I think that is it. I've also never seen it on inbound calls.
>But in my scenario, wouldn't it be rather *my* endpoint that would send
>the RE-INVITE? (for whatever reason) Since I don't have a need for
>that, I just never worried about it. So far, It Just Works. :)
Honestly, I haven't done a packet capture on it. But I don't think it's
technically a reinvite. In my scenario, I think it's just passing SDP
untouched. So the media IP's are intact and causes the switch to establish
RTP directly. (I also see the ULC's User agent (So much sonus)) This can at
times be a real pain when trying to capture the traffic on the wire simply
because you're playing Russian Roulette with which IP you're actually going
to be exchanging RTP with. But I digress. Frankly, For inbound I don't
think you'll ever see this. You could probably do something similar (No
modification of SDP) with Kamilio I suspect. But I'm not sure what this
would really do for you in your config.
>Or maybe there is even a module available for kamailio that synchronizes
>active SIP calls between multiple kamailio installations?!
>The Asterisk'es on all sites don't have anything to do with anycast,
>they just have public IPs that are local to the corresponding site, so
>they are unique.
Yeah, I meant to say Kamailio in my original email. I know asterisk
complains when it gets RTP for a call it doesn't have. No idea if Kamilio
is the same. But I imagine it would. Anycast is great for UDP based stuff
as long as you can keep it all stitched together. I'd also be curious to
see if you run into a route flap problem between a carrier and yourself at
some point. Where the route switches between your physical sites. However,
You could quickly turn some BGP knobs and mitigate that.
Thanks for sharing your Config and thoughts with the community, It's given
me some Lab Ideas!
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