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xeps/inbox/muji.xml
2015-06-24 00:39:33 +01:00

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<?xml version='1.0' encoding='UTF-8'?>
<!DOCTYPE xep SYSTEM 'xep.dtd' [
<!ENTITY % ents SYSTEM 'xep.ent'>
%ents;
]>
<?xml-stylesheet type='text/xsl' href='xep.xsl'?>
<xep>
<header>
<title>Muji</title>
<abstract>
This specification defines an XMPP protocol extension for initiating and
managing multiparty voice and video conferences within an XMPP MUC
</abstract>
&LEGALNOTICE;
<number>proto-muji0.2</number>
<status>ProtoXep</status>
<type>Extension</type>
<sig>Standards</sig>
<approver>Telepathy project</approver>
<dependencies>
<spec>XMPP Core</spec>
<spec>XEP-0045</spec>
<spec>XEP-0166</spec>
</dependencies>
<supersedes/>
<supersededby/>
<shortname>muji</shortname>
<author>
<firstname>Sjoerd</firstname>
<surname>Simons</surname>
<email>sjoerd.simons@collabora.co.uk</email>
<jid>sjoerd.simons@collabora.co.uk</jid>
</author>
<author>
<firstname>Dafydd</firstname>
<surname>Harries</surname>
<email>dafydd.harries@collabora.co.uk</email>
<jid>dafydd.harries@collabora.co.uk</jid>
</author>
<revision>
<version>0.0.0.2</version>
<date>2009-06-09</date>
<initials>sjoerd</initials>
<remark><p>Second rough draft.</p></remark>
</revision>
</header>
<section1 topic='Introduction' anchor='intro'>
Jingle <cite>XEP-0166</cite> is used to negotiate peer to peer media sessions.
Muji is a way to coordinate Jingle sessions between a group of people.
Muji conferences are held in <cite>XEP-0045</cite> rooms.
</section1>
<section1 topic="How it works" anchor="howitworks">
A Muji conference has a number of contents, each of which has unique name.
content type, and an encoding. Each participant may provide a stream for each
content, and communicates which contents they are willing to provide streams
for, along with encoding information, in their MUC presence. This serves two
purposes. Firstly, so that each participant knows which contents every other
participant provides. Secondly, so that there is a global payload type (PT)
mapping for the various contents, so that clients only need to encode and
payload each content that they provide once.
Participants are not required to participate all the contents that are
available. For example, a Muji client might choose to only request audio
streams.
</section1>
<section1 topic='Joining a conference' anchor='joining'>
<p>
Joining a conference is done in two stages. The first step is to
declare that preparations are being done to either join or start a muji
session inside the MUC. This is indicated by the client sending a presence
stanza to the MUC with a preparing element in muji section.
<code><![CDATA[
<presence from='wiccarocks@shakespeare.lit/laptop'
to='darkcave@chat.shakespeare.lit/oldhag'>
<c xmlns="http://jabber.org/protocol/caps"
node="http://telepathy.freedesktop.org/wiki/Muji"
ver="48QdBuXRCJFb8qIzgy1FOHSGO0U="
hash="sha-1" />
<muji xmlns='http://telepathy.freedesktop.org/muji>
<preparing />
</muji>
</presence>
]]></code>
The client MUST then wait until the MUC rebroadcasts its presence message,
after which it MUST wait for all other participants that had a preparing
element in their presence to finish preparation. Afterwards it should finish
it's own preparation by updating its presence with the contents it wants to
take part in.
<code><![CDATA[
<presence from='wiccarocks@shakespeare.lit/laptop'
to='darkcave@chat.shakespeare.lit/oldhag'>
<c xmlns="http://jabber.org/protocol/caps"
node="http://telepathy.freedesktop.org/wiki/Muji"
ver="48QdBuXRCJFb8qIzgy1FOHSGO0U="
hash="sha-1" />
<muji xmlns='http://telepathy.freedesktop.org/muji'>
<content name='video'>
<description xmlns='urn:xmpp:jingle:apps:rtp:0' media='video'>
<payload-type id='97' name='theora' clockrate='90000'/>
</description>
</content>
<content creator='initiator' name='voice'>
<description xmlns='urn:xmpp:jingle:apps:rtp:0' media='audio'>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
</description>
</content>
</muji>
</presence>
]]></code>
</p>
<p>
When a client adds a payload ID to a content description, it MUST have the
same codec name and receiving parameters as the corresponding entries in
other participants' payload maps for that content. For instance, if Alice
defines a payload type with ID 98, codec Speex and a a clock rate of 8000
for a content called “voice0”, then Bob must define payload type 98
identically or not at all for that content.
</p>
<p>
Furthermore, each content description MUST include at least one payload type
that every other participant supports. In other words, the intersection of
payload type mappings in descriptions for a content must not be the empty
set. This avoids clients having to encode the same stream multiple times,
which can be very costly, and also allows sending the encoded data only once
where the transport makes this possible (e.g. IP multicast).
</p>
<p>
Once a client has constructed content descriptions and advertised them in
its MUC presence, it MUST initiate a Jingle session with every other
participant. The requirement that it is the joining participant that
initiates sessions avoids race conditions.
</p>
<p>
Jingle sessions are initiated between the MUC JIDs of participants. That is,
the Jingle session-initiate stanza is sent from one MUC JID to another. This
allows participants to easily identify sessions as belonging to a Muji
conference. Content names inside Muji-related Jingle sessions always refer
to the content with the same name inside the Muji conference.
</p>
</section1>
<section1 topic='Leaving a conference' anchor='leaving'>
<p>
To leave a conference the Muji information MUST first be removed from the
participant's presence; subsequently it SHOULD terminate all Jingle sessions
related to that conference. Updating the presence first reduces the
likelihood of situations where new participants initiate sessions with
participants who are leaving the conference.
</p>
</section1>
<section1 topic='Adding a content' anchor='addcontent'>
<p>
Adding a stream follows a process similar to the joining a conference. As a
first step an updated presence stanza MUST be send which contains a preparing
element as part of the Muji section.
<code><![CDATA[
<presence from='wiccarocks@shakespeare.lit/laptop'
to='darkcave@chat.shakespeare.lit/oldhag'>
<c xmlns="http://jabber.org/protocol/caps"
node="http://telepathy.freedesktop.org/wiki/Muji"
ver="48QdBuXRCJFb8qIzgy1FOHSGO0U="
hash="sha-1" />
<muji xmlns='http://telepathy.freedesktop.org/muji'>
<content creator='initiator' name='voice'>
<description xmlns='urn:xmpp:jingle:apps:rtp:0' media='audio'>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
</description>
</content>
<preparing/>
</muji>
</presence>
]]></code>
The client MUST then wait until the MUC rebroadcasts its presence message,
after which it MUST wait for all other participants that had a preparing
element in their presence to finish their changes.
Afterwards the client should add the new content to the muji section of its
presence and add the content to all the jingle sessions it had with
participants it shared the content with.
<code><![CDATA[
<presence from='wiccarocks@shakespeare.lit/laptop'
to='darkcave@chat.shakespeare.lit/oldhag'>
<c xmlns="http://jabber.org/protocol/caps"
node="http://telepathy.freedesktop.org/wiki/Muji"
ver="48QdBuXRCJFb8qIzgy1FOHSGO0U="
hash="sha-1" />
<muji xmlns='http://telepathy.freedesktop.org/muji'>
<content name='video'>
<description xmlns='urn:xmpp:jingle:apps:rtp:0' media='video'>
<payload-type id='97' name='theora' clockrate='90000'/>
</description>
</content>
<content creator='initiator' name='voice'>
<description xmlns='urn:xmpp:jingle:apps:rtp:0' media='audio'>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
</description>
</content>
</muji>
</presence>
]]></code>
</p>
</section1>
<section1 topic='Removing a content' anchor='removecontent'>
<p>
To remove a content the participant SHOULD first sent an updated presence
without the content in its muji section. Afterwards it MUST the content from
all the jingle sessions it has open.
</p>
</section1>
<section1 topic='relays and mixers' anchor='relaysmixer'>
<p>
When scaling to conferences with a big number of participants or when
clients it's no longer viable for all participants to have direct
connections.
On connections where upstream bandwidth is the limiting factor an RTP a
RTP relay which is able to relay the stream to multiple participants on
the behalf of the clients and which forwards the streams of other
participants back to the client can be used.
If the limiting factor is either CPU or downstream bandwidth then a mixer
can be used, which receives the media streams from other participants and
mixes them on behalf of the client, so that the client only has to deal
with receiving and decoding a single stream for each media type. On the
sending side a mixer acts like a relay and relays the clients stream to all
other participants.
Both these services can either be provided by dedicated services or by
other clients.
</p>
</section1>
</xep>