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July 9 - 2010 - Issue 43
MCS Forum
An independent view of Microsoft Communications Server - Microsoft is a trademark of Microsoft Corporation |
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This week the focus is
"Vo-EYE-P" Test - Understanding Voice Sampling
For advertiser, training and special event information call Tom Cross cross@gocross.com or 303-594-1694.
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Part 1 of 10 -
Understanding and Coping with Voice QoS to Keep or Save Your Job - Avoiding the
RGE-Resume Generating Event Syndrome
The challenge
in designing any SIP/MCS or any other voice over IP network is understanding
all the things that "go bump in the night." Adding voice to a inherent unstable and
unreliable "best efforts" network is not going to let you sleep at
night or deal with the C-Level who calls your boss and you are spending time
explaining what went wrong with the "million dollar deal" call that
was lost. If you can't cope with all of
it, then
1-Get a company who can do it for you, so you can blame them,
2-Do it
yourself, not recommended or
3-Consider "no-tech" industries such as
running an winery for your new job.
According to
our research and others including Cisco:
Voice bandwidth = (Payload + L3 + L2 overhead) * 8 * PPS-Packets Per
Second (50 default)
Here is the Vo-EYE-P Test - click here for animation:
- 10 ms -
Voice Payload is 80 bytes in 10 ms-millisecond packet which means little delay
proportionally higher overhead or 72% overhead premium
20-IP + 8-UDP
+ 16-RTP + 14-Ethernet = 58 bytes overhead
- 20 MS -
Voice Payload is 160 bytes using G.711 and 20 bytes using G.729 in 20
ms-millisecond packet which means potentially higher delay but lower overhead
of 36% (160/58 overhead)
- 30 MS - Voice
Payload is 240 bytes in 10 ms-millisecond packet which means higher performance
and even lower overhead 24% but potentially higher impact of loss due to
delays.
In other words, while Microsoft, Cisco and others have suggestions, like with cars, YMMV, so be prepared to change your voice sampling just like you watch for highway traffic accidents.
In the animated tutorial you can also:
- Mouse over
for RTP-UDP-IP-Packet Overhead
- Mouse over
for Compressed Overhead
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NEW - Cisco's Tim Szigeti Speaks at T/P Expo and signs his book "Cisco TelePresence Fundamentals."
Speech Title: Best Practices: TelePresence Design
- TelePresence Technologies Overview
- Simplicity, Quality & Reliability
- TelePresence "Best Practices"
- System
- Room
- Network
Tim Szigeti, CCIE#9794, is a Technical Leader in the Enterprise Systems Engineering team at Cisco Systems. His role is to design network architectures for the next wave of medianet applications, including TelePresence, IP video surveillance, digital media systems and desktop video. He has also specialized in Quality of Service technologies for the past decade, during which he has authored many technical papers, including the Enterprise QoS Design Guide and the TelePresence Design guide. He has also co-authored the Cisco Press Books: End-to-End QoS Network Design and Cisco TelePresence Fundamentals.
Understanding and Troubleshooting Video Conferencing Networks
Tele-Presence Forum provides learning and educational services on audio, video - tele/presence, web conferencing and other aspects of tele/presence and is the sponsor of Tele/Presence Forum Expo. More information can be found at http://www.presenceforum.com |
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TECHtionary Announces training on two CompTIA certificate programs.
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Link to: Three Amigos - MS Education Gurus
Like: Where is the latest PBX compatibility list for Exchange 2010 Unified Messaging or via TechNet | |
Get SIP Smart
Proxies are signaling - Media servers are content
Click here for outline
"The SIP/VoIP courses are more than a superb primer on VoIP/SIP technology; they are an indepth business plan for a VoIP/SIP implementation. In addition, the VoIP/SIP diagnostic and troubleshooting guide is the most thorough approach to VoIP QoS available. I need information that informs but does not overwhelm. Information that guides but not drives you away. The courses provide insights and actionable information that has helped me get the technology we need sooner but saved me a considerable amount in understanding what we didn't need. The SIP course especially is a valuable one which provides much needed information in a readily understandable format." Paul Daubitz - President - ATI-TeleManagement
All Students in SIP Planning Guide Course will receive this New Cisco SIP Trunking Book (published 2-4-2010).
 SIP Trunking is one of the first complete books to planning, evaluating, and implementing high-value SIP trunking solutions. Most large enterprises have switched to IP telephony, and service provider backbone networks have largely converted to VoIP transport. But there's a key missing link: most businesses still connect to their service providers via old-fashioned, inflexible TDM-Time Division Multiplexed trunks. Cisco® authors show how to use Session Initiation Protocol (SIP) trunking to eliminate legacy interconnects and gain the full benefits of end-to-end VoIP. Written for enterprise decision-makers, network architects, consultants, and service providers, this book demystifies SIP trunking technology and trends and brings unprecedented clarity to the transition from TDM to SIP interconnects. The authors "separate the true benefits of SIP trunking from the myths and help you systematically evaluate and compare service provider offerings. This book includes detailed cost analyses, including guidance on identifying realistic, achievable savings." SIP Trunking also introduces essential techniques for optimizing network design and security, introduces proven best practices for implementation, and shows how to apply them through a start-to-finish case study. Some of the concepts explored in the book are
· Discover the advanced Unified Communications solutions that SIP trunking facilitates
· Systematically plan and prepare your network for SIP trunking
· Generate effective RFPs for SIP trunking
· Compare SIP deployment models and assess their tradeoffs
· Address key network design issues, including security, call admission control, and call flows
· Manage SIP/TDM interworking throughout the transition
· This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press - SIP Trunking book by Christina Hattingh, Darryl Sladden, ATM Zakaria Swapan was first published February 4 - 2010. |
| From MS Team MCS - helpful MCS TechTips
Click here for NextHop
Whether you're new to Microsoft Communications Server and Unified Communications, simply looking for some helpful hints, or trying to get some serious information, we'll help you find what you need. | | The GotUC.net developer sandbox is open for registration
We are a group of Microsoft Communications Server developers working together through a web-community to promote Communications Server development, to increase our skills and assist others. The founders of this site are firm believers in the potential of MCS and are committed to doing our part to make this community grow. On this site you will find many Microsoft MVPs for MCS who are willing to share their expertise with others. We are big evangelists for MCS and seek every available opportunity to speak out about it. |
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CrossTalk Named One of the Top-10 Telecommunications Blogs
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Free Technical "Just Enough Just-in-Time" Knowledge from:
The World's First and Largest Animated Library on Technology with more than 3,000 animated tutorials.
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