[Standards-JIG] Media protocol LOI

Tony Cheung tony.cheung at asiayeah.com
Wed Mar 10 12:04:06 UTC 2004


Hi Ulrich and others,

I do not know too much details about the Skype. But they have just 
announced supporting conference calling. So they have not stopped at 
only one-to-one audio and call signaling relaying.

Yes, you are right that we could think of Skype clients as supernodes 
with relaying functions. In XMPP, only XMPP servers are responsible for 
relaying XMPP messages. I understand it is for keeping the client as 
simple as possible, and the protocol do not require them to route or 
relay XMPP messages. Nevertheless, never prevents a XMPP client to act 
as a XMPP server. Right now, they probably just need to have a domain 
name on the client. If we could get rid of this limitation, we could 
easily build a more 'p2p' XMPP network for routing XML traffic. Such an 
open XML routing infrastructure may be interesting.

I am open for any discussion. It is a little bit sidetrack to the 
original Ulrich's proposal on the Media Protocol. If anyone's 
interesting in this 'p2p' discussion, we could start another thread.

Regards,
Tony Cheung

On 10 Mar 04, at 7:31 PM, Ulrich Staudinger wrote:

> Skype is, as the skype team calls it, a 3G p2p/a third generation p2p 
> network. "Supernodes" inside the closed system reroute data within the 
> system and build a dynamically changing, NAT irrelevant (due to the 
> relay nodes/supernodes) network with some lookup servers.
>
> [1] * http://www.cse.ogi.edu/webdb03/presentations/12.pdf - odissea, a 
> 3g p2p search engine.
>
> tony.cheung at asiayeah.com wrote:
>
>> Talking about voice services, Skype has gained much attention by 
>> claiming to be a p2p VoIP protocol. Is it feasible to make XMPP to 
>> provide similar features, but as an open standard?
>>
> The only skype feature that differs from our current approach is the 
> introduction of clients as supernodes. Basically you can think of a 
> RTP relay as a so called supernode.
> RTP protocol complexity handicaps upgrade of clients into RTP 
> reflectors/RTP mixers. However, in a media session we need to know of 
> and propagate media relay JIDs for conferencing anyway.
> In one-to-one environments with directly linked communication partners 
> (i.e. UDP or TCP) RTP relays are not necessary and get omitted, as do 
> the supernodes in the skype network.
>
> The differences between this half P2P and a server centric approach 
> with P2P in One-to-One conferencing are not as huge as it seems.
>
> I would not try to rebuild the skype network, to be honest, i'd 
> silently ignore the P2P term in skype and think of it as a (relay 
> system where necessary).
>
>
>
> Ulrich
>
>
>
>
>> "CORVOYSIER David FTRD/DMI/REN" <david.corvoysier at francetelecom.com> 
>> wrote:
>> __________
>>
>>> Voice and Video support through SIP is the main advantage of SIMPLE 
>>> over XMPP, so it would be a big push in favor of XMPP.
>>>
>>> However, H323 and SIP have been around for a while, and have 
>>> received a huge support in the telecom industry (all big names like 
>>> cisco, nortel or alcatel support both protocols in their routers).
>>>
>>> So you may well come up with a perfectly designed, all-purpose 
>>> merged protocol as you described below, and still be ignored by the 
>>> industry.
>>> This may not be that bad if you can count on a rich Open Source 
>>> community like Jabber however, because you will always find people 
>>> that will be happy to implement your spec just to prove that it 
>>> works better than the existing solutions.
>>>
>>> So if you decide to go for it, I would be happy to help you, based 
>>> on my (limited) experience of VoIP over XMPP.
>>>
>>> David
>>>
>>>> -----Message d'origine-----
>>>> De : standards-jig-bounces at jabber.org 
>>>> [mailto:standards-jig-bounces at jabber.org] De la part de Ulrich 
>>>> Staudinger
>>>> Envoyé : mercredi 10 mars 2004 10:32
>>>> À : Jabber protocol discussion list
>>>> Objet : [Standards-JIG] Media protocol LOI
>>>>
>>>>
>>>> We need to merge
>>>> SIP: Session initiator protocol, used for signalling, non XML
>>>> SDP: Session descriptor protocol, used to describe a session, non 
>>>> XML
>>>> SDPng: same as SDP, just in XML syntax
>>>> RTP, RTSP, RTCP: realtime transfer protocol, used to send actual 
>>>> data, non XML
>>>> H323: signalling and session descriptor protocol, non XML
>>>>
>>>> with
>>>> XMPP: IM protocol, in XML
>>>> MUC: XMPP's multi user conference protocol
>>>>
>>>>
>>>> Drawback 1:
>>>> A pure XML based approach should be favoured.
>>>>
>>>> Drawback 2:
>>>> Existing SIP and H323 routers will not natively support the new, 
>>>> merged protocol.
>>>>
>>>> Drawback 3:
>>>> The lack of a real rival to RTP/RTSP/RTCP speaks clearly for the 
>>>> use of RTP as media exchange protocol.
>>>>
>>>> Drawback 4:
>>>> No existing media (both signalling and transfer) protocol reflects 
>>>> the MMUC use cases.
>>>>
>>>> Sidenote 1:
>>>> The TINS jep merges SIP and SDPng already, but has some errors in 
>>>> the jep and  ignores MMUC.
>>>>
>>>> Conclusion:
>>>> The merged specs will be a XMPP specific protocol definition 
>>>> covering the cases of call signaling, media exchange initiation and 
>>>> media multi user conference usage.
>>>>
>>>>
>>>> Proposed solution approach:
>>>>
>>>> We use SDPng for session description.
>>>> The fundamental mechanics in SIP will be translated into SIP-XML, 
>>>> One-to-One media exchange is covered by this.
>>>> For MMUC we will simply *call* a relay, similar to dialing into a 
>>>> media conference.
>>>> All MMUC specific use cases [i.e. muting a user in a session, 
>>>> voicing on media level] will be covered within this same protocol.
>>>> MMUC will be closely related in structure to MUC.
>>>>
>>>>
>>>>
>>>> Comments requested.
>>>>
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>>>
>>
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