Global Multimedia Collaboration System

Global Multimedia Collaboration System Wenjun Wu Indiana University Bloomington IN 47401 [email_address] http:// www.globalmmcs.org

Outline Service-Oriented Collaboration Current main stream real-time collaboration technologies Videoconferencing: H.323, SIP, Access Grid Instant messaging & VoIP: MSN/Aol/Yahoo, Jabber, Skype Global-MMCS introduction

Service Oriented Collaboration Collaboration has Mechanism to set up members (people, devices) of a “collaborative sessions” Shared generic tools such as text chat, white boards, audio-video conferencing Shared applications such as Web Pages, PowerPoint, Visualization, maps, (medical) instruments …. b) and c) are “just shared objects” where objects could be regarded as Web Services We can port objects to Web Services and build a general approach for making Web services collaborative a) is a “Session Service” which is set up in many different ways

Shared Event in Collaboration All collaboration is about sharing events defining state changes Audio/Video conferencing shares events specifying in compressed form audio or video Shared display shares events corresponding to change in pixels of a frame buffer Instant Messengers share updates to text message streams Microsoft events for shared PowerPoint (file replicated between clients) as in Access Grid Using Web services allows one to expose update events of all kinds as message streams Need publish/subscrib e approach to share messages

Session Service in Collaboration Membership: Participant list Role & floor assignment Management policy based on shared objects ( audio/video, text, whiteboard, game Ad-hoc or formally schedule Session is also a “shared meta object” associated with shared objects It needs eventing to keep consistent state as well

Multimedia Streaming Service Streaming-In  Stream-Out filter process media “events” between a stream source and stream sink; and can be shared composite media service is a DAG ( directed acyclic graph) common case ~ a filter chain QoS(in)  QoS(out) ( bandwidth, delay, jitter,loss) for example: video/audio mixing, transcoding, video-audio synchronization

VoIP & Videoconferencing technology

H.323 Introduction Major audio-video standard but broader “ Binary” format for both “data” and “control” Supported by many commercial vendors and used throughout the world in commercial and educational markets Supports small-scale multipoint conferences Has conference management functionality and the call signaling functionality H.225 ~ call set-up H.245 ~ call control H.243 ~ Audio/Video multipoint control T.120 ~ Data Collaboration

H.323 Protocols H.323 is a “framework” document that describes how the various pieces fit together H.225.0 defines the call signaling and communication between endpoints (Call Signaling) and the Gatekeeper (RAS) Annex G/H.225.0 defines communication between Border Elements H.245/H.243 is the conference control protocol T.120 is the data conference protocol

Typical H.323 Stack http://www.packetizer.com H.323 IP UDP RTP RTCP TCP/UDP TCP UDP UDP TCP Audio Codecs G.711 G.723.1 G.729 .. Video Codecs H.261 H.263 H.264 .. V.150 T.120 TCP/UDP T.38 H.225.0 Call Signaling H.245 H.225.0 RAS Terminal Control and Management Data Applications Media Control Multimedia Applications, User Interface

H.323 Architecture Gatekeeper (security, QoS, routing etc.) MC MP MCU H.323 Terminal 1 H.323 Terminal 2 H.323 Terminal N .... Packet Switch Network

SIP Initially SIP was designed to solve problems for IP telephony. SIP basic functions user location resolution, capability negotiation call management. equivalent to the service H.225 and point to point part of H.245 The major difference from H.323 SIP was designed in a text format and took request-response protocol style like HTTP SIP doesn’t define the conference control procedures like multipoint parts of H.245 and T.120.

CINEMA clouds rtspd Gatekeeper SIP H.323 RTSP sipd sipum sip323 SIP-H.323 signaling gateway Conferencing Programmable SIP servers Unified messaging Streaming media Hardware SIP phone Desktop SIP clients sipgw PSTN MGCP SIP-MGCP gateway SIP-PSTN gateway Regular telephones A Integrated SIP Service System: CINEMA From Columbia University Quick-time SIPUA sipc e*phone Netmeeting e*phone sipconf

Sipconf : SIP based Centralized conferencing http://www.cs.columbia.edu/~kns10/software/sipconf SIP based conferencing server SIP/SDP and RTP/RTCP Audio mixing Play-out delay algorithm Web based conference setup G.711 A and Mu law, G.721, DVI ADPCM Multiple simultaneous conferences sipc sipconf e*phone SIP323 Netmeeting SIP/PSTN

Summary of H.323/SIP Conferencing Systems Most products are Centralized conferencing system MCU integrates the service of media processing service and session management Call-based A conference call represents control connections between clients and MCUs. MCU is just a endpoint connected to VoIP softswitch cloud Most vendors offer hardware solutions Thought as services and controllers but specialized protocols and implementations; NOT Service-Oriented Architectures! Only support small size or medium size meetings ( < 20 sites )

Access Grid Access Grid : a large scale audio/videoconference based on a multicast network provides the group-to-group collaborations among 150 nodes connected to Internet 2 world wide. Use improved MBONE audiovisual tools VIC and RAT Depends upon high-speed network ( each node needs 20Mbps ) Peer to peer architecture for distribution with centralized non standard session control (venue server) Did not develop many new capabilities but made existing public domain software better packaged and easier to use

Access Grid II Supports multiple screens and dominates some research communities

Proprietary IM MSN/Yahoo/AOL Ah-hoc small-group collaboration Text, audio-video, gaming and others…. Remote Presence Service ( typical publish/subscribe messaging application ) Massive “chat servers” running behind to support millions of users across the world Limited size of buddy list and multi-party meeting Poor/fair quality for audio/video communication Close protocol which is unfriendly to third-party developers

Skype Skype: p2p IM&VoIP solution gained a big success. improving sound quality ( use new iLBC audio codec ) from Global Sound Uses a variant of IETF Stun to identify NAT and firewall using p2p overlay (Kazaa) rather than expensive, centralized infrastructure. provided supplemental features like instant messaging service. Free on-net VoIP service and a fee-based off-net SkypeOut service that allows calling to PSTN and cellular phones Millions of download and on-line users in the world

Why is Skype so successful? Better voice quality excellent audio codec, fancy echo cancellation algorithm Global IP Sound ( iLBC audio codec ) Ability to work behind firewalls and NAT Ease of use ( quite simple UI ) based on IM metaphor P2P style without centralized MCU any peer that has enough resource can be selected to host the mixing service limited the number of participants in a conference ( at most 4 which is common for private social meetings ) use p2p overlay to discover resources and route packets

But they are simply not good enough! Although all of these systems have advantages, they are not sufficient for building more advanced and integrated collaboration systems: SIP : had a huge development recently, especially in wireless world very limited supported for conference control H.323 : AV collaboration and T.120 are not well integrated. the AV communication services and T.120 overlay networks don’t have very good scalability. H.323 and T.120 are designed in a relative complicated OSI model. It is not easy to understand and develop in their APIs Most H.323 and SIP conferencing products are based on centralized MCU And no way to take full use of private MCU resources ( Imagine how to use these private MCUs to create a meeting that have thousands of participants )

But they are simply not good enough! Access Grid heavily depends on multicast service and limited number of uni-cast bridge servers in the Internet 2 No way to be deployed in current Internet Skype : Most promising use its own propriety protocols and can’t interoperate with other legacy VoIP clients such as H.323 and SIP only support small-scale audio conferencing ( at most 5-party ) and have no video service Skype-2 is said to be able to support 10-party in dual-core Intel machines Above all, no system can support medium / large size meetings in current Internet and adapt different client devices

What’s the ideal videoconferencing system I A unified, scalable, robust “overlay” network is needed to support AV and data group communication over heterogeneous networking environments go through firewall and NAT provide group communication service in whatever unicast and multicast networks offer reliable data delivery in whatever loss network to be configured as P2P or distributed server-based overlay to provide differential services for VIP and regular users A service-oriented architecture for hosting media processing service and session control service More scalable than centralized MCU Support various style of conferencing ( massive scale of broadcasting as well as medium size of private social meetings ) Service providers can be highly distributed and p2p ~ Skype p2p audio mixing Scalable service discovery based on p2p search Customized media filters for different clients ( PC, PDA, … )

What’s the ideal videoconferencing system II A core conference control mechanism is required for establishing and managing the multi-point conference Complete conference control service like T.124 (Generic Conference Control) in T.120 framework more flexible facilities to describe application sessions and entities ( role-based, XML ) for all kinds of collaboration: audio/video, game, whiteboard Session border management Integrate different AV sessions ( H.323 , SIP, Access Grid, RealStreaming … ) Simply regard these bridging gateways as “add-on services”

Global-MMCS Service Architecture

XGSP Web Service MCU Architecture Gateways convert to uniform XGSP Messaging High Performance (RTP) and XML/SOAP and .. Use Multiple Media servers to scale to many codecs and many versions of audio/video mixing NB Scales as distributed Web Services NaradaBrokering SIP H323 Access Grid Native XGSP Admire Media Servers Filters Session Server XGSP-based Control NaradaBrokering All Messaging

Break up into “Services” Monolithic MCU becomes many different “Simple Services” Session Control Thumbnail “image” grabber Audio Mixer Video Mixer Codec Conversion Helix Real Streaming PDA Conversion H323/SIP Session/Signaling Gateways As independent can replicate particular services as needed Codec conversion might require 20 services for 20 streams spread over 5 machines 1000 simultaneous users could require: 1 session controller, 1 audio mixer, 10 video mixers, 20 codec converters, 2 PDA converters and 20 NaradaBrokers Support with a stream optimized Grid Farm in the sky Future billion way “Video over IP” serving 3G Phones and home media centers/ TV ’s could require a lot of computing

GlobalMMCS and NaradaBrokering All communication – both control and “binary” codecs are handled by NaradaBrokering Control uses SOAP and codecs use RTP transport Each stream is regarded as a “topic” for NB Each RTP packet from this stream is regarded as an “event” for this topic Can use replay and persistency support in NB to support archiving and late clients Can build customized stream management to administer replay, and who gets what stream in what codec NaradaBrokering supports unicast and multicast Use firewall penetration and network monitoring services in NB to improve Q0S

NaradaBrokering Stream NB supports messages and streams NB role for Grid is Similar to MPI role for MPP Queues

Incorporating Support for Audio/Video Delivery into NaradaBrokering Added support for an unreliable transport protocol, UDP Implemented a fixed size (fast) topic (8 bytes). Designed a new compact event with minimum headers. Added support for legacy RTP clients (both unicast clients and multicast groups) Improved the routing algorithm to handle real-time audio and video stream delivery.

XGSP Conference Control Architecture

XML based General Session Protocol The XGSP conference control includes three services: Conference management supports user sign-in, user create/terminate/join/leave/invite-into XGSP conferences conference calendar service Application session management provides users with the service for creating/terminating application sessions, managing session related services such as audio/video mixing Floor control manages the access to shared collaboration resources in different application sessions for example, in a large scale of meetings having thousands of people, only limited people are allowed to become presenters so that they can send audio/video

Global-MMCS Community Grid This includes an open source protocol independent Web Service “MCU” which will scale to an arbitrary number of users and provides support for thousands of simultaneous users of collaboration services. The function of A/V media server is distributed using NaradaBrokering architecture. Media Servers mix and convert A/V streams Open XGSP MCU based on the following open source projects openh323 is basis of H323 Gateway NIST SIP stack is basis of SIP Gateway NaradaBrokering is open source messaging Java Media Framework basis of Media Servers Helix Community http:// www.helixcommunity.org for Real Media http://www.globalmmcs.org open source release

Audio/Video Meeting Tests for single broker Audio Meeting Tests Video Meeting Tests

Distributed Brokers Tests Going through multiple brokers does not introduce considerable overhead. Scalability of the system can be increased almost linearly by adding new brokers

Analysis of the broker network’s performance Test results showed that the broker network can scale well for both single large size meetings and multiple smaller size meetings. In large size meetings, the capacity of the broker network is increased with respect to the capacity of the added brokers. In multiple smaller size meetings, the distribution of users among brokers are important. Inter-broker stream exchange can reduce the scalability. Few users should not be scattered around the broker network. In wide area networks, this videoconferencing system provides many benefits with distributed broker architecture: bandwidth savings, latency savings, and better quality services. In summary, thousands of concurrent users can easily be supported in distributed broker settings.

media services computation overhead 90% while 4, 23fps stream producers are running Real Streaming Producer 70% while 50 image grabbers are running Image Grabber 94% while 4 video mixers ( 4-way mixing ) are running Video Mixing 46% while 20 audio mixers ( six active speakers ) are running Audio Mixing Computation Overhead Media Services

Improved JMF Performance Fraction of CPU used versus number of received streams The CIF-size video sequence from a 30-second movie with a lot of motions is streamed to the clients. Each stream is encoded in H.261, and has average bandwidth of 400Kbps and 20 fps. 34% - 35% 62% - 63% 40% - 41% 8 31% - 32% 58% - 59% 35% - 36% 7 27% - 28% 51% - 52% 32% - 33% 6 23% -24% 46% - 47% 26% - 27% 5 17% - 18% 40% - 41% 23% - 24% 4 15% - 16% 33% - 34 % 17% - 18% 3 9% - 10% 24% - 25 % 13% - 14% 2 6% - 7% 15% - 16 % 8% - 9% 1 Fast JMF Client Old JMF Client VIC

Polycom, Access Grid and RealVideo views of video-mixed streams using GlobalMMCS

MPEG-4 vs. H.261 We added MPEG4 video to Java Media Framework Higher quality and flexible video sizes including distributed pixels

Coupled Diverse Streams GlobalMMCS supports many diverse streams managed by “video system” Different audio and video codecs Shared display using video codecs (MPEG4 or H261) Motion JPEG – stream of images to and from PDA NaradaBrokering represents these and other collaborative streams just a “topics”; collaboration from multiple clients subscribing to a topic Text Chat Traditional lossless codec based shared display White boards Control streams Streams can be linked to provide composite topics eSports project linking video streams and real time annotation of any frame Can rewind and choose any frame of a real-time stream

eSports Snapshot Master Video Annotation Whiteboard Collaborative Video Annotation Whiteboard Synchronized Replay of archived video and annotation

Integration of PDA, Cell phone and Desktop Grid Access NB Support for optimized PDA Communication

GlobalMMCS Status/Futures I 1. New Collaboration tools Shared IDL (Visualization), PowerPoint, OpenOffice (Applications need a month or so more) SVG game ( stable ) Whiteboard ( stable ) e-Sport ( prototype) Jabber IM client ( prototype) XGSP needs extension to support 2. JMF Audio/Video client ( stable) performance enhancement finished new codec ( MPEG4-DivX finished; try MPEG4-Xvid and H.264) support different platform ( Linux, Mac – Mac well developed but need to chase bug(s) ) support NAT/firewall transparently like Skype Google Desktop PlugIn (under development )

GlobalMMCS Status/Futures II 3. Replay & Archive (prototype) Replay Engine based on NaradaBroker Storage Service XGSP-RTSP gateway Extend RTSP and NaradaBrokering for Instant Replay 4. Web Server Portal ( stable) Standard calendar service ( iCalendar, vCalendar) Flexible conference management Need to package UI’s as portlets 5. Conferencing Media Processing Service ( Stable) Support new codec (H.264 ) Add DCT domain MPEG4-H.261 transcoder 6. H.323 Gateway ( Stable) Import it to Linux platform

GlobalMMCS Status/Futures III 7. RealStreaming Gateway ( Stable ) Import it to Linux Support Mobile clients 8. Global-MMCS deployment & test Core performance measurements complete Test under the setting of multiple NaradaBroker and NAT/Firewall support deployment for AFRL, NASA, DOE portals test with remote sites 9. SmartPhone Clients (prototype) 10. Improved video codec-based shared display 11. S cheduler of dynamic services sensitive to streaming bandwidth requirement as well as CPU use of codec conversion

Global Multimedia Collaboration System

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