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Re: [MMUSIC] BUNDLE: no reason por payload type to be unique within bundled m= lines

Iñaki Baz Castillo <ibc@aliax.net> Thu, 17 March 2016 09:38 UTC

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References: <CALiegfm7Rb4e5DFGycr=EsdigE1XFCUiqNL6+GpRSyrGe=_RWg@mail.gmail.com> <56DED735.1000101@ericsson.com> <CALiegf=G77JOcoR6DORnPsMx4OCgy=_dYQ_r8KfjMy_33vn89A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37E9EA74@ESESSMB209.ericsson.se> <CALiegf=F4z9r+oCO1DMW_nXq7TBJjf-n3N1r6yr5tLoksXtUqg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37E9FDAB@ESESSMB209.ericsson.se> <CALiegf=kzThQ3n+Z-=E8G=ivyL1W=Q4gUcQenihAJ3pmv8udKQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37EA257B@ESESSMB209.ericsson.se> <CALiegfnLMUZnA=AQEnvFhxKKbwUeg=sFHRiC9R2MxisY-43UFg@mail.gmail.com> <D31037D8.5B90%christer.holmberg@ericsson.com> <CALiegfmWPrai9a1hkj5rpLeyowspAcpr3jNZpNyc4HatXGfo9Q@mail.gmail.com> <D3103E3F.5BBE%christer.holmberg@ericsson.com>
From: Iñaki Baz Castillo <ibc@aliax.net>
Date: Thu, 17 Mar 2016 10:30:30 +0100
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE: no reason por payload type to be unique within bundled m= lines
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2016-03-17 10:09 GMT+01:00 Christer Holmberg <christer.holmberg@ericsson.com>:
>>I just hope that BUNDLE is a required step in order to separate each
>>RTP "sender" and RTP "receiver" within a RTP session, so in the future
>>we can get rid of SDP and design a proper RTP parameters based API for
>>WebRTC.
>
>
> That is not the purpose of BUNDLE. The purpose is to be able to negotiate
> multiplexed media on a single RTP session

- "RTP session" == stuff carried over a single transport (let's assume
rtcp-mux always).

- "media" == audio / video / data, which can be sent and/or received.

So yes, BUNDLE somehow allows defining each RTP sender and/or receiver
in a separate m= section.


> If we want to change the rule saying that PT values must be unique per
> codec/codec configuration, I think that¹s a separate task.

We can't do that because there are a lot of legacy and very expensive
RTP middleboxes made in 20th century that miraculously are suitable
for WebRTC even if they don't even know what a SSRC is. And they must
work without any code update!


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

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