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

Iñaki Baz Castillo <> Mon, 03 October 2011 12:26 UTC

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Date: Mon, 3 Oct 2011 14:29:24 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <>
To: "Olle E. Johansson" <>
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Subject: Re: [rtcweb] Future requirement: RTC-Web apps must work through interface switching
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2011/10/3 Olle E. Johansson <>et>:
> 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.

> 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.

> 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.


Iñaki Baz Castillo