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sip failover?
The situation is the following. The local network. There is a telephony server on the network. It was decided to temporarily move the operators working through this server to another office. And leave the server in place. What we currently have. Two subnets (office and remote office). Mikrotik here and there. And there and there are two channels to the Internet from two different providers. For each channel, I will launch a vpn-tunnel connecting mikrotiks. And now the question in the studio). How to implement full operator connection failover... without frame loss? There is an idea to wrap packets in both tunnels and reject one at the exit, and put the second into operation. One of the tunnels is down ... just use the packets that come from the second tunnel. But how to implement it? Or do you have better ideas?
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I would do channel balancing, and transfer packets would be assigned to the tcp / ip stack
"How to break protocols designed to survive frame loss so that they work without frame loss"
Well, turn on RTP over TCP, you will have a conversation without frame loss (even Wikipedia says that this is a perversion).
In telephony, it is more profitable to lose packets, from the point of view of human hearing.
Even if a scheme similar to the one described is implemented, a serious problem will arise with two channels - the delay between packets in the channels. Those. we need a buffering mechanism to synchronize data in the channels to make sure that we lost / did not lose packets, and then restore their correct sequence and only after that transfer to the PBX. This will result in serious audio delays, most likely beyond what is comfortable for a conversation. And in this case, it is easier to transfer the scheme to TCP with greater reliability due to the simplification of implementation, but exactly the same delay problems.
such a question is worth a million - it’s somehow not interesting to make an SD-WAN project for a like.
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