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SIP (Session Initiation Protocol) is an IETF standard multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks.
SIP is an open standard replacement for the ITU's H.323.
SIP is described in RFC 3621 - SIP: Session Initiation Protocol.
SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.
SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.
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In the previous discussion on QoS the Per-Hop Behaviors DiffServ uses to mark packets were identified.
These where listed as:
- Expedited Forwarding (EF) – RFC 3246 – Provides a strict priority service
- Assured Forwarding (AF) – RFC 2597 – Provides a qualified delivery guarantee, and provides for oversubscription, markdown and dropping schemes for excess traffic
- Class Selectors (CS) – RFC 2474 – Provides code points that can be used for backward compatibility with IP Precedence models
- Best-Effort – Provides delivery when possible
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In part one of this series on QoS we determined that QoS was the differential treatment of the voice, video, and data packets that flow on the IP network, creating a system of managed unfairness. QoS technologies allow different types of traffic to contend inequitably for network resources. Time-sensitive applications, such as voice or interactive video packets, can be given priority over data applications.
In this post we’ll examine the three different models of QoS that are supported in Cisco IOS software.
Click Here - To read the rest of this blog on the Global Knowledge Web Site
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QoS ensures more predictable network services by providing dedicated bandwidth, controlled jitter and latency, and improved loss characteristics. QoS provides tools for managing network congestion, shaping network traffic, using WAN links more efficiently, and setting traffic policies across the network. QoS helps provide consistent, predictable network performance by offering intelligent network services.
With all of the above being said, it is no wonder so many students feel uneasy with the subject of QoS. In this series of blogs I will take a closer look at the many functions of QoS.
Click Here - To read the rest of this blog on the Global Knowledge web site
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Call Admission Control (CAC) is often times included as part of the same topic as Quality of Service (QoS), when in actuality CAC is a separate and complete topic itself.
QoS is defined as traffic engineering on a packet switched network. This definition means moving the IP Packets on to the wire and across the network in the fastest time possible, with the least amount of dropped packets. QoS manages this process by prioritizing different data flows. Packets with high sensitivity to the amount of time it takes to traverse the network receive a higher priority; such as voice and video packets.
Click Here - To read the rest of this blog on the Global Knowledge web site
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During a recent client upgrade from CM 4.13 to CUCM 7.x, it came to my attention that CUCM (Cisco Unified Communication Manager – formerly CallManager) 7.x did not ship with AC (Attendant Console).
My client, and many others, use the attendant console feature in their CM 4.x network, and plan to continue the use of AC in their upgraded CM network.
It also came to my attention that many Cisco integration partners do not know that AC does not ship with CM 7.x and are not including a plan in their upgrade design to allow for this issue.
Here is what I found in researching how to deal with the issue of AC not shipping with CUCM 7.x.
Click Here - To read the rest of this blog on the Global Knowledge web site
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Call Admissions Control (CAC) prevents over-subscription of the VoIP network by counting the number of calls on a network. Acting like a network traffic cop, CAC disallows VoIP calls to traverse a network when the network has reached its maximum number of VoIP calls allowed. CAC mechanisms complement the capabilities of Quality of Service (QoS) tools to protect voice traffic from the negative effects of too many voice streams on an oversubscribed network.
Click Here - To read the rest of this blog on the Global Knowledge web site
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When integrating a Voice over IP (VoIP) system into an existing network it is very important to have a good understanding of how much bandwidth is utilized for each call on the network. For most people, just starting out the bandwidth calculations can be a very daunting task.
But, with a basic understanding of the components of a VoIP packet, the process of calculating the VoIP bandwidth becomes easier to understand. It is very important to understand the bandwidth allocation for VoIP packets on the network. If underestimated this could lead to latency issues and poor application performance.
Click Here - To read the rest of this blog on the Global Knowledge web site
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Over the next few years as IP Telephony and VoIP take over the communications industry, many companies will be faced with the difficult decision to migrate their antiquated telephony infrastructure to VoIP. Cisco Unified Communications (UC) combines all forms of business communications into a single, unified system that provides powerful new ways to collaborate. MeetingPlace is a significant component of the suite of products Cisco Systems calls “UC”.
Cisco Unified MeetingPlace Release 7.0 is a Session Initiation Protocol (SIP)-based architecture that provides voice, video, and web conferencing solution to large enterprises. Housed within the customer’s private network, it allows for a lower cost of service and maximum security. End users can enter MeetingPlace conferences by using a variety of desktop applications such as IM clients, IP Telephones, and web browsers. Because of its capabilities to utilize IP Based phones, PSTN, groupware clients and web browsers, MeetingPlace users can ensure their participation and availability in conferences from any location.
Click Here - To read the rest of this blog on the Global Knowledge web site
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Unified Communications (UC) combines all forms of business communications into a single, unified system that provides powerful new ways to collaborate. Encompassing several communication systems or models, UC includes unified messaging, collaboration and interaction systems, real-time and near real-time communications, and transactional applications. The termination of Private Branch Exchanges (PBX) and Public Switch Telephony Networks (PSTN) circuits to be transported across an IP network and delivered to another phone system is traditionally referred to as Voice over Internet Protocol (VoIP). This IP-based solution would be the driving force behind the obsolescence of traditional phone switching equipment at the customer site. This new technology would be known as IP Telephony.
Click Here - To Part 1 of this blog on the Global Knowledge Web Site
Click Here - To read Part 2 of this blog on the Global Knowledge web site
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Just a few short years ago people had never heard of unified communications (UC). Today, at the top of most IT department’s priority list is UC. Due to the combination of productivity improvements and cost savings that UC can deliver to the corporation, most companies are considering a UC roll-out or UC pilot sometime in the near future.
Click Here - To see full blog on Global Knowledge Web Site
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