Re: [rtcweb] STUN for keep-alive - RTCP-less applications

Iñaki Baz Castillo <ibc@aliax.net> Thu, 22 September 2011 11:05 UTC

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Date: Thu, 22 Sep 2011 13:08:21 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [rtcweb] STUN for keep-alive - RTCP-less applications
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2011/9/22 Hadriel Kaplan <HKaplan@acmepacket.com>om>:
>> All of the above are clear barriers against legacy interoperability.
>
> Yup.  I don't doubt we'll need some form of media-plane gateway either in the web server or separate to interop with legacy - I'm just trying to keep it from becoming so expensive and complicated that it would be cheaper/easier/better to use SIP softclients or plugins instead.

I can understand that some requirements can be imposed to RTCweb
environments in the media plane, but try at least to make it
compatible with VoIP networks out there. For example, requiring SRTP o
ZRTP could make sense (anyhow I don't see why that should be a
requirement in a local intranet in which plain-RTP is already used for
SIP communication).

If WebRTC forces the creation of media gateways when
intercommunicating with other RTP worlds (SIP, XMPP) that's not good
at all. Adding requirements it's ok, making it impossible by design is
not.

Regards.


-- 
Iñaki Baz Castillo
<ibc@aliax.net>