[OpenSIPS-Users] Call continuity

Thu Jul 20 16:07:28 EDT 2017

Hi Everyone,

This is a non-related OpenSIPS question, it is a SIP question.

I am not expert in SIP , so let me know what you guys think about this.

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?

Thanks

Abdirahman Osman

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