[OpenSIPS-Users] Call continuity

Mike Tesliuk mike at wsu.com.br
Sat Jul 22 15:41:11 EDT 2017

This is the kind of structure where you need a BGP session with your
carriers, when you have an ASN and your own IP Block you can have your
communication flowing between any internet links you want, as the ip
address will flow between all of them, this is the right way.

if you have two simples internet connections there is no way to do that

Em 20/07/17 17:11, Alex Balashov escreveu:
> On Thu, Jul 20, 2017 at 08:07:28PM +0000, Abdirahman A. Osman wrote:
>> Is it possible to keep a live call continue , if the internet
>> connection fails and route it through another internet connection ?
>> Does SIP protocol support this kind of Call continuity?
> Generally speaking, no, though the answer will vary with the modalities
> of the failover mechanism.
> But in general, failing anything over to another Internet connection
> means changing the address of one of the endpoints involved. All
> session-based Internet applications, whether using a
> connection-orientated transport or not, presume that the IP and port
> endpoints on both ends stay the same. 
> So, if you suddenly start sending media from another place and expecting
> to receive it there likewise, that will not be considered to be part of
> the same phone call.
> -- Alex


Mike Tesliuk
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