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Update on SIP Conferencing

Update on SIP Conferencing. SIPPING WG IETF 60 San Diego, CA August 6, 2004. Documents. draft-ietf-sipping-conferencing-requirements-00.txt draft-ietf-sipping-conferencing-framework-01.txt draft-ietf-sipping-cc-conferencing-04.txt draft-ietf-sipping-conference-package-05.txt

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Update on SIP Conferencing

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  1. Update onSIP Conferencing SIPPING WG IETF 60 San Diego, CA August 6, 2004

  2. Documents • draft-ietf-sipping-conferencing-requirements-00.txt • draft-ietf-sipping-conferencing-framework-01.txt • draft-ietf-sipping-cc-conferencing-04.txt • draft-ietf-sipping-conference-package-05.txt • See also URI list documents

  3. Updates to cc-conferencing • Added definition of conf option tag for UAs • Added requirement and flow for deletion of conference • Added scenario of participant requesting focus refer a participant to the conference • Added scenario of moving a point to point call to a conference with a separate focus

  4. Planned Changes to Conf Package • Structure of Current Draft: • User • Status/joining mode • Media • Instance • Structure in Future: • User • Session (instance) • Status/joining mode • Media • Content • Move the following elements to a new XCON Conf Package Extension draft • Media stream label • Policy URIs • Add hooks for known and future XCON extensions • With these changes, draft will be ready for WGLC

  5. Example of New Structure <?xml version="1.0" encoding="utf-8" ?> <conference-info version="0" state="full" entity="sip:conf233@example.com"> <user uri="sip:bob@example.com" display-name="Bob Jones"> <role>"participant"</role> <session id="4fdfdfdf" instance="sip:bob@client.example.com"> <status>connected</status> <joining-mode>dialed-in</joining-mode> <media media="audio"> <proto>RTP/AVP</proto> <ssrc>583398</ssrc> </media> <media media="video"> <proto>RTP/AVP</proto> <ssrc>645839</ssrc> </media> </session> </user> <user uri="sip:barbara@example.com" display-name="Barbara Jones"> <role>"participant"</role> <session id="9uytgfre" instance="tel:+13145551212"> <status>connected</status> <joining-mode>dialed-in</joining-mode> <media media="audio"> </media> </session> <session id="fderes6sd" instance="sip:barbara2@client.example.com"> <status>connected</status> <joining-mode>dial-out</joining-mode> <media media="video"> <proto>RTP/AVP</proto> <ssrc>8233398</ssrc> </media> </session> </user>

  6. To Do • Move conf option tag text from cc-conferencing to a SIP document • Finalize split of conference package • Finalize split with URI list documents • Conference Factory URI definition/flows will move from cc-conferencing to a new SIPPING draft which includes INVITE URI list draft. • Then, all four documents ready for WGLC

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