[Standards-JIG] Media protocol LOI

Ulrich Staudinger us at die-horde.de
Wed Mar 10 11:31:11 UTC 2004


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.
>>>
>>>_______________________________________________
>>>Standards-JIG mailing list
>>>Standards-JIG at jabber.org 
>>>https://jabberstudio.org/mailman/listinfo/stan> dards-jig
>>>
>>>      
>>>
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>>--- message truncated ---
>>    
>>
>
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