Voip-performance Management and Optimization - Download as PDF File .pdf), Text File .txt) or read online. 'VoIP Performance Management and Optimization' Excerpt. KB | 3 files | null DOC, null PDF. This book chapter will give your team an overview of issues. The definitive reference for managing and optimizing VoIP networks on VoIP network performance management Analyzes both reactive and List Price: $; Includes EPUB, MOBI, and PDF; About eBook Formats.
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viii VoIP Performance Management and Optimization. Contents at a Glance. Foreword xx. Introduction xxi. Part I. VoIP Networks Today. Chapter 1. Voice over IP. VoIP performance management for a converged network data quality. Our solutions also apply to various steps in the assess, monitor, manage and optimize. Aberdeen's research shows that the top pressures driving enterprises to focus resources in managing VoIP performance are: • The need to optimize cost of.
Vertical bars separate alternative, mutually exclusive elements. Square brackets [ ] indicate an optional element.
Braces within brackets indicate a required choice within an optional element. Comcast came in on top at 5.
This puts great emphasis on managing VoIP for both enterprises and service providers. The first stage is planning, which includes forming a team, defining the scope of deployment, requirements validation, and assessment of the IP network to determine whether the infrastructure is adequate to support media traffic.
The design phase includes comprehensive design-based traffic engineering and validated requirements. It not only covers call-processing servers, remote gateways, and features implementation but also covers changes to the IP network in the form of quality of service design and provisions for network resiliency. It is followed by the implementation phase, which is governed by project management principles and ensures that best practices for deployment are followed for on-time completion.
Implementation also includes a test plan execution and transfer of information to network operations prior to commissioning. The correct execution of these phases ensures minimum problems and decreases the total cost of deployment.
Implementation is followed by the operations phase, with continuous optimization to close the loop. This book briefly mentions planning, design, and implementation stages and emphasizes the operations and optimization phases. First, the hand-off to operations needs to be complete, including remediation of any issues discovered when the postdeployment test plan was executed. All the deployed devices must be discovered by the network management systems.
But most important, VoIP can no longer be managed in a silo that is separate from the data network management subteam. This book emphasizes correlating network problems with VoIP-related key performance indicators for faster problem resolution by isolating it and fixing the root cause.
Operational data provides critical feedback for continuous optimization of the network, including its performance and capacity. Optimization is not limited to fine-tuning the traffic engineering process for future growth but also for extending VoIP for the next evolution to collaboration-enabled business transformation. What is presented in this book is the authors collective experience and knowledge, working with several other colleagues from Advanced Services, Cisco Remote Operations Service, the product development teams, and most important, Cisco customers, whose feedback was critical in developing best practices for VoIP management and optimization.
Customers not only need to fix problems in a timely manner with minimal downtime, but they also need to proactively monitor their networks to fix potential problems before they become service and revenue impacting.
The complexity of an IP network increases with the addition of new services, and as these networks start to scale, managing them becomes a challenge.
Customers are looking for new ways to manage their networks and effectively scale these services. This feedback can be boiled down to We want a practical guide with specific details and examples that we can use right away This book addresses some of the challenges associated with deploying and managing VoIP networks and also provides guidelines on how to optimize these networks.
Goals and Methods The most important goal of this book is to help define a methodology and framework of collecting, analyzing, and correlating VoIP performance data from various network elements. When correlated in a meaningful way, this data can help network operators identify problematic trends in their VoIP networks, and isolate and fix problems before they become service impacting.
One key methodology in this book is to use a layered approach when troubleshooting VoIP network problems.
Voice over IP
This helps narrow the scope of the problem in an efficient manner and also helps find the root cause. By quickly identifying the root cause of the problem, the network operator can resolve issues in a timely manner and minimize customer impact. This book also provides guidelines for optimizing VoIP networks by defining the following: What VoIP performance data should be collected from various network elements?
How to collect VOIP performance data? How to use dashboards to analyze and correlate VoIP metrics? Phone calls between subscribers of the same provider are usually free when flat-fee service is not available. These are typically designed in the style of traditional digital business telephones.
An analog telephone adapter connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cablemodems have this function built in.
Softphone application software installed on a networked computer that is equipped with a microphone and speaker, or headset.
The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input. VoIP switches may run on commodity hardware, such as personal computers.
Rather than closed architectures, these devices rely on standard interfaces. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi network, so that it is no longer necessary to carry both a desktop phone and a cell phone. Maintenance becomes simpler as there are fewer devices to oversee. Two kinds of service providers are operating in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business SMB market.
It is a best-effort network without fundamental Quality of Service QoS guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to data loss in the presence of congestion [a] than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.
Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. Latency can be minimized by marking voice packets as being delay-sensitive with QoS methods such as DiffServ. Excessive load on a link can cause congestion and associated queueing delays and packet loss. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. VoIP endpoints usually have to wait for completion of transmission of previous packets before new data may be sent.
Although it is possible to preempt abort a less important packet in mid-transmission, this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. But since every packet must contain protocol headers, this increases relative header overhead on every link traversed. Packet delay variation results from changes in queuing delay along a given network path due to competition from other users for the same transmission links.
VoIP receivers accommodate this variation by storing incoming packets briefly in a playout buffer , deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout , i. Although jitter is a random variable, it is the sum of several other random variables which are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question.
Motivated by the central limit theorem , jitter can be modeled as a gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful.
In practice, the variance in latency of many Internet paths is dominated by a small number often one of relatively slow and congested bottleneck links. Most Internet backbone links are now so fast e. RFC VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications. This is generally down to the poor access to superfast broadband in rural country areas.
With the release of 4G data, there is a potential for corporate users based outside of populated areas to switch their internet connection to 4G data, which is comparatively as fast as a regular superfast broadband connection. This greatly enhances the overall quality and user experience of a VoIP system in these areas. A virtual circuit identifier VCI is part of the 5-byte header on every ATM cell, so the transmitter can multiplex the active virtual circuits VCs in any arbitrary order.
Cells from the same VC are always sent sequentially. Every Ethernet frame must be completely transmitted before another can begin. If a second VC were established, given high priority and reserved for VoIP, then a low priority data packet could be suspended in mid-transmission and a VoIP packet sent right away on the high priority VC. Then the link would pick up the low priority VC where it left off. Because ATM links are multiplexed on a cell-by-cell basis, a high priority packet would have to wait at most 53 byte times to begin transmission.
There would be no need to reduce the interface MTU and accept the resulting increase in higher layer protocol overhead, and no need to abort a low priority packet and resend it later. Some examples include: IEEE The standard is considered of critical importance for delay-sensitive applications, such as voice over wireless IP. MTA connects the subscriber equipment. VoIP access is provided at the customer premises. Application Servers: These servers include voicemail VM servers for providing voice mailbox service to subscribers.
These are used for billing purposes. It establishes a physical connection with the CMTS and forwards traffic between the SP network and the subscriber equipment. Media Gateways MGW: By plugging a standard analog telephone into the MTA device. Figure reflects such a topology. These protocols allow a central call control mechanism to control customer premises equipment CPE devices for voice services.
They achieve this by placing a trunking VoIP switch. The architecture presented in Figure touches on various technologies.
VoIP Performance Management and Optimization
It can be a collection point for MGCP-related metrics. The MGCP-based communication counters for announcement servers. This is also called a voice gateway that utilizes E1 lines in the case of non-U. The CMS acts as a central switching point and is thus an ideal place for collecting key performance indicators KPI because it acts as central switching component.
Based on the SIP. SIP Proxy Server: Cisco Access Servers: The Cisco access gateway provides universal port data. PGW is also called a trunking switch. The Cisco PGW is a carrier-class call agent that performs the signaling and call-control tasks such as digit analysis. Customers can take advantage of the many new IP telephony applications by connecting their analog devices to Cisco ATAs. The traffic is routed through SIP proxies onto the trunking gateway.
This shows the SIP connectivity from various sources: Figure represents a mixture of networks with various integration boundaries. Note the various integration points and the diverse protocol networks. PSTN incoming and outgoing calls.
Figure also shows trunk connectivity between the two PGW gateways that are respectively part of large. In this particular architecture. The other major use of IP trunks is across international boundaries. To manage the VoIP service. The SIP. The signaling and other key metrics can help in tracking.
They can be tracked by signaling and media metrics and can help in sizing and service-level assurance SLA for the integration points. All the services provided by these networks need to be tracked. Another solution for a Call Control Server Farm is also shown in the figure.
The counter collection points are the respective switching. The PGW keeps track of all CDRs and is extensively used to apply policies for routing traffic through it to optimize cost.
This offloading is provided by a switch that is performing Class IV or toll-switch functionality. Chapter 7 covers these KPIs in detail.
The corresponding traffic counters represent the KPIs needed to effectively monitor the network. This section discusses some of the common-use cases to continue the discussion of VoIP networks and the corresponding components. All these networks are shown to be carrying different types of traffic along with possibly the VoIP traffic. In VoIP networks. Figure shows SIP. That way. Other usage includes voice transcoding and network hiding. Figure also shows various SBC deployment scenarios.
In general. The call and protocol metrics provided by the switch are crucial for running the VoIP network in an efficient way. SBCs are used at the edge of the network. Topology Hiding. Topology Hiding CAC. VPN Interconnect. CCM Interworking. The downtime for this kind of service has many repercussions. Effective monitoring of the circuits through KPIs allows quick resolution of the outages.
Both of these call bearer and signaling aspects have their own infrastructure.
The PSTN trunk turnup procedure requires countless hours of interaction with the telco. The VoIP SP has to constantly monitor the capacity and continue to profile the traffic to keep up with the subscriber growth.
The needs or challenges for the VoIP SP that drive this design are numerous and include the following: The main theme here is to provide a background of the various operational overheads. The monitoring through select KPIs allows effective operations. As you will see in Chapters 7 and 8.
Another key challenge is to keep on top of SIP trunk utilization and call performance metrics. Maintenance is another challenge.
The endpoints that are connected to the edge of the SP network should be authenticated using the Authentication. Being able to look into the traffic enables the SBC to perform a wide range of functionality. Configuring access control lists ACL on the edge routers can help prevent unwanted traffic in the network. To summarize. This functionality is offered by all SBCs. The collection. It covers the security challenges seen at the network element and discusses what can be done to address them.
Hiding allows the SP to not expose its internal network to the outside work. The key concept behind black-holing is to stop the propagation of this kind of traffic. These security measures can include deploying stateful firewalls in the network that allow only authorized traffic to enter the SP network. This section provides more context for this discussion. SDP is basically used for multimedia session setup. Unauthorized users should be denied access to network resources by either black-holing their traffic or assigning them a low bandwidth class of service that would not allow them to send or receive a significant amount of traffic.
A security association is a set of provisioned security elements for example. The endpoints in the conversation can negotiate a set of ciphersuites type of authentication and encryption to be used and then encrypt all their traffic using the negotiated method.
To do this. The issues are ongoing and need to be maintained for the life of the VoIP service. Recall that a converged network is defined as a network capable of transmitting all types of traffic including data. The interaction that takes place between the endpoint and the trusted device can be encrypted.
VoIP is primarily deployed on converged IP networks. This ensures that all traffic between the two devices is from known sources and encrypted. Although these groups have totally different job responsibilities and technical background.
The VoIP traffic is sensitive to time. In the case of a centralized switching model. By having a set of security associations. Most existing SP IP networks have been designed to carry primarily data traffic and are geared toward data applications such as email.
IP Security IPsec is one of the mechanisms used to achieve this with the preprovisioned preshared keys. Convergence-Related Issues As discussed earlier in the chapter. Other voice quality issues can be caused by things such as codec mismatch.
Excessive packet loss can cause issues such as choppy voice quality.
VoIP Performance Management and Optimization
These issues are often caused by interoperability issues between equipment from different VoIP vendors. Voice signaling—related issues can also be caused because of improper network design and misconfigurations. If voice signaling gets impaired. These routing protocols carry network information that is used for calculating the most efficient path for carrying customer traffic through the SP network.
VoIP traffic or packetized voice traffic needs to be sent at fixed intervals at the transmitting end so that the receiving end can predictably receive these packets and decode them. If the dejitter buffers overflow because of excessive delay. Network congestion and resource oversubscription can adversely affect voice-signaling protocols that can affect the services and features these protocols support. In some cases.Then the link would pick up the low priority VC where it left off.
This chapter also covers the monitoring mechanism available to network administrators and their scope and effectiveness in managing VoIP networks. An application programming interface API is a source More information. A More information. A security association is a set of provisioned security elements for example.
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