Network Ninja

There are numerous benefits to packet switched telephony:

• More efficient use of bandwidth and kit: Traditional telephony networks use a 64-kbps (For argument lets say 1B Channel on a ISDN line) channel for every voice call. Packet telephony shares bandwidth among multiple logical connections and offloads traffic volumes from existing voice switches.
• Lower costs for telephony network transmissions: A substantial amount of equipment is needed to combine 64-kbps (ISDN) channels into a high-speed link for transport across a network (Lets say an ISDN PRI). Packet telephony statistically multiplexes voice traffic alongside data traffic. This consolidation represents substantial savings on CAPEX and OPEX.
• Consolidated voice and data network expenses: Data networks functioning separately from voice networks become major traffic carriers. The underlying voice networks can be converted to utilize the packet-switched architecture to create a single integrated communications network with a common switching and transmission system. The benefit is CAPEX and OPEX savings.
• Increased revenues from new services: Packet telephony enables new integrated services, such as broadcast-quality audio, unified messaging, and real-time voice and data collaboration. These services increase employees productivity and profit margins well above those of basic voice services. In addition, these services enable companies and service providers to differentiate themselves and improve their market position.
• Greater innovation in services: Unified communications use the IP infrastructure to consolidate communications methods that were previously independent (Fax, voicemail, email, wireline telephone, cellular phone, and the web). The IP Infrastructure provides users with a common method to access messages and initiate real-time communications – independent of time, location, or device.
• Adding to new communications devices acket technology can reach devices that are largely inaccessible to the time-division multiplexing (TDM) infrastructures of today (pcs, wireless devices, household appliances, PDAs). Access to these devices enable companies and service providers to increase the volume of communications they deliver, the breadth of service they offer, and the number of subscribers they serve. Packet technology, therefore, enables companies to market new devices, including videophones, multimedia terminals, and advanced IP Phones.
• Flexible new pricing structures: Companies and services providers with packet-switched networks can transform their service and pricing models. Because network bandwidth can be dynamically allocated, network usage no longer needs to be measured in minutes or distance. Dynamic allocation gives service providers the flexibility to meet the needs of their customers in ways that bring them the greatest benefits.

The basic components for voice on a IP network are as follows:

• IP Phones: The end-device on desks
• Gatekeeper: Provides Connection Admission Control (CAC), bandwidth control and management and address translation.
• Gateway: Provides translation between voice over Internet Protocol (VoIP) and non-VoIP networks, such as the public switched telephone network (PSTN). It provides physical access for local analog and digital devices (telephones, fax machines, and PBXs)
• Multipoint Control Unit: Provides real-time connectivity for participants in multiple locations to attend the same videoconference or meeting.
• Call Agent: Provides call control for IP Phones, CAC, bandwidth control and management, and address translation.
• Application Server: Provides services such as voicemail, unified messaging, and Cisco CallManager Attendant Console.
• Videoconference Station: Provides access for end-users participation in videoconferencing. This station has a video camera and a microphone. The user can view video streams and hear the audio that originates from the remote user station.

There are other components not listed here like voice applications, interactive voice response (IVR) systems, and softphones that meet the specific needs of enterprise.

Voice and Data Traffic Characteristics

Voice traffic has extremely stringent QoS requirements (because it is extremely delay sensitive). Voice traffic generates a smooth demand on bandwidth and has minimal impact on other traffic (60 – 120 bytes), as long as voice traffic is managed. Because of the resulting time sensitive nature User Datagram Protocol (UDP) is used to package voice packets; TCP retransmit capabilities have no value (because if it needs to be retransmitted then there is delay in the actual conversation occuring NOW).

For voice quality, delay should be no more than 150ms (one-way) and less than 1% packet loss. A typical voice call requires 17 – 106 kbps of guaranteed priority bandwidth, plus additional 150bps per call for voice-control traffic. Multiplying this out for the maximum calls expected during busiest times the overall bandwidth requirements for voice traffic can be calculated.

Because Data traffic is not as delay sensitive and can tolearate high drop rates the restransmit capabilities of TCP has become important, as a result many applications use by default TCP.

In networks, important business critical applications are ussually easy to identify. Most applications can be identified based on TCP or UDP port numbers (HTTP, HTTPS, FTP, TELNET, SQL, ETC). Some application use dynamic port numbers that, to some extent, make classification more difficult. Cisco IOS software supports network-based application recognition (NBAR), which can be used to recognize dynamic port applications.

VoIP Call Flow

As I mentioned in a previous post (see HSRP Accross Trunk Links) and some other places its best practice to setup voice and data on separate VLANs (I did in my own network). This is done so that QoS can be applied to prioritize the VoIP traffic as it traverses the network. If this is not done then voice and data traffic contend for available traffic without consideration for other devices (one or the other is going to suffer).

A major component of designing a successful IP Telephony network is bandwidth provisioning. The bandwidth requirement is calculated by adding the total required bandwidth for voice, video and data together; the sum should not be more than 75% of the link total.

For a traffic perspective IP Telephony consists of two types of traffic:

1. Voice Carrier Stream consists of Real-Time Transport Protocol (RTP) packets that contain actual voice samples.
2. Call Control Signaling that contains packets belonging to one of several protocols used to set up, maintain, tear down, or redirect calls. Depending on the end-point this could be H.323 or Media Gateway Control Protocol (MGCP)

Auxiliary VLANs

Some Cisco Catalyst switches offer a unique feature called “Auxiliary VLAN“. This feature allows one to overlay a voice topology over an existing data network. One can segment phones into a separate logical network, even though the data and voice network are physically the same.

The auxiliary VLAN feature places the phones into their own VLANs without any end-user configuration. Additionally VLAN assignment can be maintained even if the phone is moved.

How this works is that when a phone is plugged into the switch (whichever port), the phone will request a DHCP address, and the phone is placed in a VLAN automatically. With phones in their own VLANs administrators can troubleshoot and identify problems easily. This also makes enforcement of QoS and security policies easier.

QoS

QoS is the application of features and functionality required to actively manage and satisfy the networking requirements of applications that are sensitive to loss, delay and delay variations (jitter). QoS allows preference to be given to critical application flows for the available bandwidth.

Cisco IOS implementations allows for QoS to provid these features:

• Priority access to resources: QoS allows administrators to control which traffic it allows to access specific network resources such as bandwidth, kit, and WAN links.
• Efficient management of network resources: If network management and accounting tools indicate that specific traffic is experiencing latency, jitter, and packet loss, then QoS tools can be used to adjust how traffic is handled.
• Tailored service: The control provided by QoS enables Internet Service Providers to offer carefully tailored grades of service to their customers.
• Coexistance of mission-citical applications: QoS technologies ensure that mission-critical applications receive priority access to network resources while providing adequate processing for applications that are not delay sensitive.

High Availability

Traditional Telephony networks strive to provide 99.999 (5.25 minutes) of downtime a year. This is less downtime than most data networks. To provide the same experience this means choosing hardware and software with a low mean time between failure (MTBF) or installing redundant links and hardware.

Availability is when a user wants to make a call the network is able to respond to that need. Efforts to ensure availability would include proactive management to predict failure and taking steps to correct problems in design of the network as it grows. When the converged network goes down things downtime can be minutes, hours or days. This is unacceptable in a converged network where downtime means no phone calls. Providing for uninterpretable power supplies (UPS), lighting arrestors and other means to ensure availability at all costs.

High Availability encompases many areas of a network. In a fully redundant network these components need to be duplicated:

• Servers and call managers,
• Acces layer devices (layer-2 switches)
• Distribution layer devices (routers or Layer-3 switches)
• Core layer devices (layer-3 switches)
• Interconnections (WAN links, PSTN Gateways, ISP links)
• Power supplies and UPSs

Notes and Notices:

This is a part of my personal BCMSN notes and research to assist myself in learning and understanding the concepts and theory for the BCMSN exam. I learn by making notes reading and writing things down and wish to file them where I can’t lose them. These notes are not to be seen, judged or mistaken for replacements to Cisco recognized and authorized training which I personally support and attend and suggest you undertake if you are going for the BCMSN Certification.