RC2 is a block cipher designed by Ron Rivest in 1987 (Rivest Cipher)
Firewall / NAT Traversal is a potential road block to deployment Extranet collaboration (via the DMZ) Internal separation of divisions The answer is standards Control signaling and media must be considered separately For control signaling, SIP proxies must be strategically located to act as application-layer routers for all endpoints For A/V media ICE/STUN/TURN standards come to the rescue New Sametime components in Sametime 8.5.x ICE/STUN/TURN support built into Sametime 8.5.x A/V clients and servers New TURN server component (similar function to Sametime Reflector in 8.0.2)
Sametime 8.5.1 and earlier Leverages existing network tools to identify/manage different media packets types to follow network QoS rules Sametime AV packets are tagged to allow identification by QoS system Sametime 8.5.x controls use and protects network bandwidth New feature of Sametime Media Manager Call admission control based on location in network Protects network and constrains bandwidth usage Call succeeds, fails, or is renegotiated (e.g., video to audio only) Define “Class of users” and location call rate policy Based on defined logical network topology model
Sametime 8.5 introduced SIP based Media Manager Enables native audio/video conferences between Sametime clients 3rd party extensions enable interop with audio/ video conference bridges e.g., Polycom, Radvision, iLink (Tandberg), Avaya, PGI, Avistar, others
The capabilities in each category are shown here … more detail on each is available in following slides
This graphic shows the user experience of a number of the Sametime client integrations … IP Communicator Click to Call with Lotus Sametime Cisco Phone Control with Lotus Sametime Cisco Phone Presence with Lotus Sametime Unified Messaging with Lotus Sametime WebEx Click to Conference with Lotus Sametime Instant Messaging
As a rule of thumbs, for calculating bandwidth, assume one concurrent lotus sametime video client call for every 10 users. Assume half of these calls would be on a bridged call. This is the current snapshot of Polycom conference platform. Polycom 1500 is a closed box so there is no hardware expansion. For RMX 2000 and RMX 4000, you can add media modules to increase system’s capacity. For typical deployment, Lotus Sametime users (computer) will be connected at CIF resolution. (Louts Sametime supports up to 1080P high definition, but it requires both powerful CPU and bandwidth. Most typical deployment for hundreds or thousands of computer video users is CIF-equivalent. (352 x 288) Polycom RMX can manage resources dynamically, which is to say, consume only the resources required to support the resolution used. Therefore, if there are more computer video users than room video conference system with HD video, then the resources are optimized so that more computers can be connected simultaneously.
IBM Sametime 8.5.2 Audio / Video Vincent Perrin | IBM Certified Collaboration Solutions Architect June, 2011
Internal Users Only - No NAT Alice Bob Community Server Media Servers Meeting Server ST Proxy Server
Internal Users Only – with Firewall / NAT Alice Bob Community Server Media Servers Meeting Server ST Proxy Server TURN Server NAT NAT
DMZ – Option I Internal Users (LAN) DMZ External Users (Internet) Alice Bob Community Server Media Servers Meeting Server ST Proxy Server TURN Server
DMZ – Option II Internal Users (LAN) DMZ External Users (Internet) Alice Bob Community Server Media Servers Meeting Server ST Proxy Server TURN Server SIP Proxy Edge Server HTTP Reverse Proxy Community Mux
Bandwidth Manager provides network capacity (“bandwidth”) control and protection via class-of-user and location-based call rate policies and network topology modeling
The back-office administrator can control access to available bandwidth on each segment of the network topology by provisioning call rate policies to limit the network bandwidth for audio and video data
The policy constrains the amount of bandwidth available for audio and video so that audio and video calls won't interfere with other traffic on the network
Policy is associated with sites or groups of sites in the network topology, specific users or classes of users, or specific predefined groups of users
The administrator can monitor usage of bandwidth in order to tune topology and policy settings
Message Flow (relationship to SIP Proxy) 1. The message is first sent from endpoint A to the SIP Proxy 2. The SIP Proxy routes the message to BWM according to the routing rules 3. BWM inserts its Contact URL and modifies the SDP payload based on the applied policies then sends the message back to the SIP Proxy 4. The Proxy Registrar sends the modified message to endpoint B
The endpoints can be either the Media Manager Conference Focus, Packet Switcher, or any of the supported clients depending on which component initiated the SIP flow
All subsequent messages in the same SIP dialog follow the same route
The SIP Proxy avoids endless looping since the Contact header indicates that BWM server has already been visited
INVITE sip:B From: sip: [email_address] To: sip: [email_address] Contact: sip:<IP A> payload SDP Endpoint A INVITE sip:B From: sip: [email_address] To: sip: [email_address] Contact: sip:<IP BWM> payload SDP - modified 1 2 3 4 Endpoint B SIP Proxy Bandwidth Manager
SameTime Server Web Services Virtual Places (VP) Protocol ACE Server (Standalone or High Availability) Application Integration Engine (AIE) Server Virtual Places (VP) Protocol TR87 & SIP JTAPI & SIP TR87 & SIP SIP Media Application Server (MAS) (NMC) TCSPI Avaya ACE and IBM Integration ACE Server, ACE TCSPI and AIE to support IBM Desktop CS1000 CISCO CUCM Avaya AES/CM Tandberg VCS
RMX 2000 RMX 4000 RMX 1500 Resolution Resources Voice 1440 CIF/HD VSW 360 SD/4CIF 240 HD 720p 120 HD 720p 60fps 60 HD 1080p 60 Resolution Resources Voice 720 CIF/HD VSW 180 SD/4CIF 120 HD 720p 60 HD 720p 60fps 30 HD 1080p 30 Resolution Resources Voice 360 CIF/HD VSW 90 SD/4CIF 60 HD 720p 30 HD 720p 60fps 15 HD 1080p 15 ~ 1800 Sametime Users* ~ 3600 Sametime Users* ~ 7200 Sametime Users* *Assume 256k or 384kCIF desktop video. Assumes one concurrent Lotus Sametime video client call for every 10 users. Assume half of these calls would be on a bridge. This will need to be adjusted due to Lotus Sametime architecture, time zones diversity and other. factors Conference Infrastructure Scale Scale Resources and Redundancy
Chris: Accept the call from ST client Peter: Accept call from HDX in the pull-down list
RMX calls out to HDX (h.323, etc) and ST clients (SIP) dial in
Seamless Escalation from IM to Multiparty Video
Use Case 1: Click-to-conference Select contacts and 1) start chats and escalate to video call 2) directly start a video call Lotus Sametime users receive incoming call notification and select device from which to receive the call. Users can simply add Polycom RMX hosts the voice and video multiparty calls with continuous screen for rich conference experience
Use Case 2: Web Collaboration Enter a Sametime meeting room (ad hoc or scheduled). Invite others (drag & drop contact names into Meeting Room or choose contacts from Meeting Room UI) The recipient of the meeting joins the meeting room. Participants dial into the Polycom RMX from within the meeting room window
Plantronics Plug-In for IBM Sametime The late st Hands free headset call control between IBM Sametime and Plantronics’ UC audio devices, provide users with exceptional online meeting experiences with enhanced audio quality and connectivity including: 1. Control Sametime calls remotely from the Headset: Answer/end, mute and volume control features allows you to gain hands-free mobility directly from the audio device if you need to roam away from your desk, or are multi tasking whilst on Sametime /SUT Calls. 2. Put Sametime calls on hold remotely from the headset : Put your Sametime Caller on hold so you can conduct business privately and know your caller is secure and your alternative conversation is confidential 3. Switch between Sametime calls remotely from the headset: Answer other inbound calls including Skype calls whilst multi tasking away from the computer. Don’t miss out on connecting with colleagues and customers needing to speak with you simultaneously. 4. Smart Sensor Wearing State: Answers call automatically as soon as headset is (donned) or placed on the ear without the need to press ‘answer’ button on headset -or on the screen. 5. Mobile telephony presence enhances both Sametime and SUT by eliminating the ‘blind spot’ in SUT by changing Sametime ‘presence’ status to ‘On Phone’ when a user is on a mobile telephone / cell call. This requires no user intervention as the headset solution automatically detects the mobile phone call and changes the presence status in Sametime
Jabra, an IBM Business Partner, seamlessly integrates unmatched endpoint audio quality with IBM® Sametime® and Sametime Unified Telephony. Jabra headsets deliver an enhanced user experience, enabling users to fully control Sametime softphones from the headset while providing hands-free freedom and mobility.
Exceptional sound quality using the latest advances in audio technology combines with ergonomic design to promote productivity and user satisfaction.
The Sennheiser Call Control plug-in surfaces key functionality within the Sametime user experience by providing a seamless integration of the Sennheiser headset with IBM Sametime.
You benefit from a fully tested solution making use of IBM Sametime together with Sennheiser headsets. All call-control features can be used via the Sennheiser headset, i.e. answering and ending calls, adjusting volume or muting the call. The DW Series wireless headsets provide you with maximum mobility to roam around the office while still being able to communicate.
crop Vincent Perrin Lotus Collaboration Solutions Architect IBM Software Group 17, avenue de l'europe Bois Colombes Tel +33 677 02 03 54 [email_address]