Re: [rtcweb] Future requirement: RTC-Web apps must work through interface switching

["Olle E. Johansson" <[e-mail]>]

3 okt 2011 kl. 14:29 skrev Iñaki Baz Castillo:

> 2011/10/3 Olle E. Johansson <[e-mail]>:
>> THe SIP outbound extension has handling of IP address change during a registration/connection. But that doesn't imply IP address change during a call.
> 
> When a SIP client implementing the outbound extension (RFC 5626)
> detects that the TCP (or TLS or WebSocket) connection is
> terminated/down, it should re-register and should send a re-INVITE for
> established dialogs telling the peer about its new location and SDP
> address. In fact, the using RFC 5626 the client does not need to know
> its exact local IP:port, as the edge proxy would route back incoming
> in-dialog request using the new client connection.
Over UDP the client recognizes a new binding using STUN and determines that the connection has failed
and re-initializes.

> 
> 
>> Software like Asterisk has added protection on the RTP ports so after initial packets (handling symmetric RTP) Asterisk will not accept packets from an unknown IP.
> 
> It should allow a change in the source IP:port of the RTP packets in
> case a re-INVITE has been sent by the peer.
Always. Of course. But previously we always accepted it...
> 
> 
>> I would assume that if we had ICE support, ICE could re-initialize the session and the server could open new connections.
> 
> I expect this to be the same: the client would send a re-INVITE after
> detecting local address change and Asterisk should accept it and
> update the session information.
> 
Right.

/O