|
|
|
September 2- Issue 51
| 
MCS Forum is an independent forum on
Microsoft Communications Server & UC-Unified Communications.
Microsoft is a trademark of the Microsoft Corporation.
|
|
|
Our Sponsors
|
Visit Our Sponsor
Call 303-594-1694 for more on Vimenture
Click here mention PromoCode MCSTEL - ask for John Cognata
SOFTEL Communications Inc. specializing in the development and delivery
of best-in-class IP telephony, unified communications, contact centers,
and speech self-service solutions. From simple single site to complex
multi-channel, multi-modal, multi-platform, enterprise implementations,
SOFTEL provides full life-cycle professional services, consulting,
design, application development, solutions integration and
implementation services that focus on reducing operational costs and
improving business performance.  Click here or call 303-594-1694  Wireless Directory Assistance Patent Opportunity
Click here or call 303-594-1694 |
|
|
Greetings!
Welcome to the MCS Forum.
A new look for a new era in unified communications. In you would like to see your products highlighted, reviewed and presented here or in other publications such as:
|
|
QoS Series - Part 1 - Jitter: Causes & Cures
Let's review Causes of Jitter - Congestion
QoS is critical to business VoIP as any congestion can cause
"stutter" delays, "voice skips" from packet loss or
"jitter" in the conversation. In a "two-box" environment,
the premise router can only look to the cable modem but not "look
ahead" to see network congestion which limits QoS. Data networks are called "bursty"
networks because data arrives randomly and in large amounts often causing
"traffic jams." A "one
box" environment enhances QoS by making "smart decisions" about
prioritizing network traffic. Adding bandwidth is just one way to reduce
voice/data "log jams" and VoIP "stuttering" problems. Advanced routers or other devices such as
network optimizers and equalizers are hardware software devices that give the
network administrator control over every type of packet and TCP port.
The animation details all the processes. Click here to view, note it moves pretty fast, so if you are too old or just hate animation or simply don't understand the internet, then you may want to just read the text.
The
security features of many routers include capabilities such as hardware-based
virtual private network (VPN) acceleration, firewall and intrusion
prevention. These and other techniques
are critical as video, gaming, video mail and other complex systems such as
virtual reality emerge.
Cures
One of the key advantages of new routers is providing
both a traditional T-1 circuit and a "metro" cable ethernet solution
for routing. Either circuit can be the
primary routing circuit, however, in case of circuit failure; alternate
circuits can be selected automatically.
In addition, protocols like HSRP-hot Standby Routing Protocol can be implemented
for more protection.
Network equalizing
is another choice in balancing of traffic allowing for time/delay-sensitive
voice, video and priority data to receive an "equal share" of the
available bandwidth. Network equalizing
is a dynamic QoS process which can respond to changes in network
conditions. In addition, as more and
more applications are placed on the network, network equalizing is critical to
"filling the pipe" and not paying for idle bandwidth (see animation for complete details).
More Cures in the next issues - Clock Synchronization and Clock Drift - Jitter Buffers and more
|
QoS Series - RTCP-Real Time Control Protocol
This section is included to educate viewers on the key elements in RTCP and to have a QoS strategy in place BEFORE implementing SIP/OCS to avoid an RGE.
NOTE: IF YOU CAN'T READ THE SLIDES CLICK HERE
NOTE: Click here for the complete details on the RTCP-XR-MRB.
RTCP-XR-Real Time Control eXtended Reports is one of the key tools in diagnosing and
troubleshooting VoIP networks. XR-eXtended Report technology can be integrated into IP Phones or PSTN
Gateways. XR packets (see graphic for
packet format) are sent periodically during the call to provide real time
feedback on call QoS-Quality of Service. However, VoIP/SIP network planners need to consider the amount of XR
traffic that also consumes and reports on voice traffic. That is, diagnostic XR traffic consumes
bandwidth to diagnose traffic. The RTCP
MRB-Metrics Report Block provides measurements (metrics) for monitoring quality
of VoIP calls and conversations. These measurements include packet loss and
discard metrics, delay metrics, analog metrics, and voice quality metrics. The
Metrics Report Block reports individually on packets lost (discarded) on the IP
channel as opposed to packets that have been received and then lost by the
receiving jitter buffer. MRB reports on the combined effect of losses and
discards which can be used to determine corrective actions on voice QoS.
Thanks to AudioCodes for their help in this presentation. |
New OCS 2007 R2 Cumulative Update 6 available
Link to: Three Amigos - MS Education Gurus
|
Replay of Cisco SBC and MTP - Redundancy-Resiliency & Scalability
Click here for animated tutorial
From the "must-read" book VoIP Performance and Optimization from click here Cisco Press ISBN 1-58705-528-7, the authors Ahmed, Madani and Siddiqui present, "The Session Border Controller (SBC)or the border element (BE) can provide another level of address masking while performing other tasks such as call filtering, call normalization (CODEC interoperability and fast-start and slow-start call setup methods) and bandwidth management. SBC/BE simplifies interoperability by allowing only one conduit to be opened for access to the aggregation point."
"The Cisco Unified Communications Manager (CUCM) and all the endpoints, including IP phones and gateways have private IP addresses. The SBC/BE (Border Element) and the MTP-Media Termination Point have public addresses. MTP and SBC functionalities can be offered in one physical device. Also, there can be several MTPs and/or SBCs for redundancy-resiliency and scalability. All the media from the SP-Service Providers network are sent through the MTP. There is no direct connectivity between the IP phones, Unified CM and outside the world." They also discussed theft of service, involving using network resources to place long distance calls that incurred high told charge or exploiting the resources by inter-trunk transfer can also occur elsewhere in the book.
Two Types of SIP Offer Invites
with SDP-Session Description Protocol
- Early Offer-Fast Start Invite
- SDP is sent with the Invite (advertises its CODEC/media capabilities,
encryption and other terms of call)
- Delayed Offer-Slow Start
Invite - Invite is sent without the SDP (called party advertises CODEC/media,
etc.) The "Offer" typically defines the media characteristics supported by the
device (media streams, CODECs, directional attributes, IP address, and ports to
use). The Offer Invite is contained in
the Session Description Protocol fields sent in the body of a SIP signaling
message. The SIP endpoint receiving the Offer sends an "Answer" in the SDP
fields of its SIP response, with its corresponding matching media streams and
codec, whether accepted or not, and the IP address and port on which it wants
to receive the media streams. Details on SDPs can be found in RFC-3261. Note: In either case, codec/media selection by
either called party or calling part in not unilateral decision but rather a
negotiation.
If an MTP or SBC is involved in
the either invite process, they can also act as "proxy servers" to negotiate
terms of either fast or slow start invitations.
Some reasons for using Early
Media include:
· The called device might want
to establish an Early Media RTP path to reduce the effects of audio cut-through
delay (clipping) for calls experiencing long signaling delays or to provide a
network-based voice message to the caller.
· The calling device might want
to establish an Early Media RTP path to access a DTMF-Dual Tone Multi-Frequency
or voice-driven IVR-Integrated-Interactive Voice Response system.
|
"One
Softphone Size Can Fit Nearly All"
One Softphone for Nearly ALL Platforms
"One Size Can Fit Nearly All"
IS-phone is a second generation voice over IP telephony
application, which offers IP telephony (audio/video) features smoothly
integrated into nearly all platforms - PBX to Web. It enables corporate
telephony (coupled to your IP PBX), presence, audio/video conferencing and
advanced click-to-dial from your Lotus contacts or LDAP directory, buddies,
emails or documents. is-phone provides key advantages of simplicity,
flexibility and attractive pricing exposed by pure software solution.
IP-PBX certified on IBM, Avaya, Broadsoft, Apple
AdTran/Objectworld, Alcatel-Lucent, Asterisk, BroadSoft, Cisco, Comverse,
MetaSwitch, Mitel, NEC Sphere, Nortel, OpenSIPS, Siemens, SIP Express Routers,
SIPfoundry etc. If your platform is not listed, click below and I will check on
it. In addition, a web page solution is available without regard to any
browser. These are just two examples.
Click here with
PromoCODE - MCSISO and will get you even more. |
Domestic & Global Network Services
Bandwidth and other Communications Services
Vimenture has teamed with TECHtionary to provide data,
internet, conferencing and other network services including:
- MPLS-Multi-Protocol Label Switching
- Ethernet - Metro
- Hosted Servers - Cloud Computing
- Telecommunications Bill Audits
- Security Assessments
- International Connectivity
- Voice Over IP - SIP Trunking
- Conferencing Services - Audio
- Professional Services - Planning, Integration, Consulting
Click here and
mention
PromoCode OCSVIM. |
MCS Knowledge Source
Click here for Next Hop from MS -- "Whether you're new to Office 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."
|
TECHtionary Announces training on two
CompTIA certificate programs.
For complete details click here.
|
Onsite SIP Course Get SIP Smart
"Proxies are signaling - Media servers
are content"
For SIP Course details click here.
 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.
|
Book Review
"VoIP Performance Management and Optimization"
By Adeel
Ahmed, Habib Madani,
Talal
Siddiqui.
For Cisco Press details click here.
There is really nothing about
this book I don't like except that it is only on paper, not electronic but read below and you get access to it online. That
is the only negative think I can say about this book and if you are serious
about VoIP (SIP) performance, QoS, security, monitoring and infrastructure
integration (hardware) and more, then you need to read and know everything that
is in this book. And like what my mom
said to me a long time ago, "if you think you know what's going on, then
you are really full of s##." Seriously there is just too much really
good information to mention in less than 300 words (trying to keep in
brief). Here's one of my favorites: on
page 262 "signaling traffic is also vulnerable to attack, including Spam over Internet
Telephony-SPIT. SPIT leverages SIP proxy
impersonation to sent unsolicited bulk messages to SIP endpoints. VoIP phishing (vishing or fishing) involves
CallerID spoofing and then call rerouting to dummy IVR systems for further exploitation
of the SIP call processing resources." This one of the many great "actionable" tutorials you will find in the
book.
With your book
purchase you are entitled to free, instant online access to that book on Safari
Books Online for 45 days. After you've completed your purchase, you will
receive instructions on how to log into Safari Books Online. If you do not want
to receive online access to the book, simply uncheck the box for Instant Online
Access in your cart.
|
Sales Performance Improvement
 Cross Passes Microsoft Partner Network Sales and Marketing Competency Assessment for Unified Communications 
helping improve sales (job aids, marcom),
proven performance and results are always
available click here.
|
TECHtionary Knowledge Source
Click here for TECHtionary -- World's First and Largest Animated Library on Technology with more than 3,015 animated tutorials.
|
|
|
|
|
|
|
|
|
MCS
Forum provides classroom and webseminar training as well as a non-production
environment for those IT departments without additional equipment, budget or
time. This allows planners and users to test ideas, dial in and dial back
out, IM file transfers, remote desktop sharing, video conferencing, run
scenarios, review logs, break linkages and learning about new telephony
features and network access. MCS Forum is also designed for both the
system integrator/consultant who wants to learn about MCS without having to
build their own system as well as the enterprise customer who doesn't have the
time, resources or knowledge to develop one. MCS Forum Labs are
designed to be "hands-on" or "over-the-shoulder" with
experts available for Q&A and classes for feature-specific review.
About MCS Forum
MCS Forum is a vendor-independent
laboratory environment designed for learning, technical guides, knowledge
resources and online "live" services. MCS Forum provides
planning, project management, consulting, training, case studies, white papers,
speaking engagements, market/customer research, network planning and other
services.
Click here to contact.
|
|
|
|
