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<body style="overflow-wrap:break-word; word-break: break-word;"><div class="mail_android_message" style="line-height: 1; padding: 0.5em">Dear all,<br/><br/>See the e-mail below.<br/><br/>I had to update the vision about "Simple Multiuser over SIP" (SMUoS).<br/><br/>Have a nice week<br/>Christoph<br/><br/>--<br/>Diese Nachricht wurde von meinem Android Mobiltelefon mit GMX Mail gesendet.</div><div class="mail_android_quote" style="line-height: 1; padding: 0.3em"><html><body>Am 18.02.24, 10:32 schrieb Christoph Valentin <christoph.valentin@gmx.at>:</body></html><blockquote class="gmail_quote" style="margin: 0.8ex 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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Dear All (at BCC),
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Dear Friends of Realtime 3D,
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Dear Friends of Telecom,
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Dear Friends of Mission Critical Networks,
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What, if we combined the protocols DIS and SIP? What, if we combined X3D with SIP? -> see the mail below
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Is this idea actually new?
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Well, it is not new at all.
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It is very old.
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I issued an idea report "IMS + X3D" to the Siemens Innovation Management, some 15 years ago (must have been 2008 or 2009 or so). Markus helped me, those days, and actually we are stilll obliged to keep our mouths shut.
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Now, after 15 years, it would be slightly weird, if we still kept our mouths shut. Therefore we founded an association, last July, to be able to help to test the development of that idea.
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However, we don't own enough resources to be able to implement the idea on our own, we are dependent on your help.
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But, what exactly was the idea?
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The idea was, to use the functions of the IMS (which is the "Telekom SIP Implementation"), in order to improve X3D multiuser scenes.
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<li>store the avatars at the HSS</li>
<li>use the authentication, authorization and accounting of the IMS, in order to improve X3D multiuser scenes</li>
<li>The "shared state" of X3D MU scenes is nothing else, than yet another media type within the SDP, like "voice" and "video"</li>
<li>The position of the avatars could be used in the MGW (MRF), in order to influence the loudness of the voice channels (so called "3D voice chat")</li>
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This is it,
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Have a nice day
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Yours Christoph
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CP/V - Christoph Peter VALENTIN
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<b>Gesendet:</b> Samstag, 17. Februar 2024 um 17:46 Uhr
<br> <b>Von:</b> "Christoph Valentin" <christoph.valentin@gmx.at>
<br> <b>An:</b> "Valentin Christoph" <Christoph.Valentin@kontron.com>
<br> <b>Betreff:</b> FYI (Verein "The Society"): Vision slightly updated
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Dear all,
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<strong>This is a private information</strong>. <strong>No connection to our company</strong>.
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I felt free to update the "Vision of SMUoS (Simple Multiuser over SIP)" at our web site.
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Reason: I think it is better, if the SIP clients sends ALWAYS to 224.0.1.75, but not to 239.255.x.x.
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Here the link:
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<a href="https://lc-soc-lc.or.at/sub-topic-001-smuos/vision.htm">https://lc-soc-lc.or.at/sub-topic-001-smuos/vision.htm</a>
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In the P.S. the updated text:
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<p>Have a nice week,</p>
<p>Yours Christoph</p>
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<p>P.S.: the updated text</p>
<p>So we should <b>implement a very simple SIP client that interfaces with the used Web3D browser</b></p>
<p>Better: can we use some <b>existing SIP client?</b> The goal of "The Society" is not to implement anything, but to test 3rd party software. Only, if we need something very special to test something else, then we will implement something.</p>
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<li><b>First step</b>:
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<li>1.1. no REGISTER, only INVITE, <b>no SIP server at all</b></li>
<li>1.2. INVITE is directly sent from SIP client to mcast address 224.0.1.75 "all SIP Servers" (sip:room-name@224.0.1.75)</li>
<li>1.7. all clients that know the room-name answer with 200 OK</li>
<li>1.8. ACK is sent to each client via unicast address from Contact header field</li>
<li>1.9. no provisional responses</li>
<li>1.10. only 200 OK as possible answer (or no answer at all)</li>
<li>1.11. ACK. BYE.</li>
<li>1.12. Contact header field holds name of the user plus the unicast address of the client</li>
<li>1.13. Question: how to describe DIS session in SDP?</li>
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<li><b>Second step</b>:
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<li>2.1. no REGISTER, only INVITE, <b>simple SIP server</b></li>
<li>2.2. the same as 1.2.</li>
<li><b>2.3. simple SIP server (new element) listens to mcast address 224.0.1.75 "all SIP servers"</b></li>
<li><b>2.4. answers with 302 Redirect (redirects to 239.255.x.x multicast address)</b></li>
<li><b>2.5. SIP client sends second INVITE towards 239.255.x.x (same Call-ID?, same From-Tag?)</b></li>
<li><b>2.6. mapping from room name to mcast address 239.255.x.x is configured at the kamailio server</b></li>
<li>2.7 - 2.13. the same as 1.7 - 1.13.</li>
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<li><b>Third step</b>:
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<li>3.1. <b>REGISTER, INVITE, complex SIP Server</b></li>
<li>3.2. - 3.5 the same as 2.2. - 2.5.</li>
<li><b>3.6. some kind of "admin" must authenticate at kamailio an can then register a 239.255.x.x address for the name of a "chat room" dynamically (mapping need not be configured at the kamailio server)</b></li>
<li>3.7 - 3.13. the same as 2.7 - 2.13.</li>
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