[VoiceOps] Ideas for Building Inbound Redundancy

[Nick Olsen]

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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