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[SLIDE 1 – title slide]

VoIP—What is it?

[SLIDE 2-VoIP—what is it?]

Voice over Internet Protocol (VoIP)

VoIP enables the traditional networking infrastructure to carry voice traffic over an IP network. In VoIP, a digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with the International Telecommunications Union-Telecommunications (ITU-T) specification H.323, the specification for transmitting multimedia (voice, video, and data) across a network.

Drivers behind the convergence between voice and data networking (see p.19-20 of VoIP book):

 Data has overtaken voice as the primary traffic on many networks built for voice  The PSTN cannot create and deploy features quickly enough  Data/Voice/Video cannot converge on the PSTN as currently built  The architecture built for voice is not flexible enough to carry data

Circuit-switched calls require a permanent 64Kbps dedicated circuit between two telephones—the connection can’t be used by any other party whether it is in use or not, and the phone company cannot use this bandwidth for any other purpose, so must bill the parties for consuming its resources. Data networking, on the other hand, has the capability to only use bandwidth when it is required. This is a major benefit of packet-based voice networking.

[SLIDE 3—DYA]

[SLIDE 4—buzz words]

The phrase “Multiservice networking” is used a lot as is “convergence.” These both imply a strategically important issue for enterprise infrastructures. This idea is essentially the combination of all types of communications--data, voice, and video--over one physical infrastructure. The major benefit is reduced operational costs—there is only one physical infrastructure to deploy and maintain.

Other “Enterprise applications” and benefits:

 Toll-bypass (moving intra-office voice and fax calls over an existing TCP/IP network and OFF the PSTN—large long-distance charge savings for typical businesses)  “Click-2-dial” links on web pages  MS Netmeeting—integration between traditional phone services with application sharing and H.323-based videoconferencing.  IP Phones  Cisco’s PC-based soft phone—extends the handset functionality on to the PC with a GUI that provides the same functionality as the handset and integrates with other multiservice applications such as web browsing, netmeeting, or directory services based on LDAP. It also eliminates the need to have an additional device (i.e., a handset) on each desktop as the soft phone utilizes headsets and speakers. [SLIDE 5—IP Telephony benefits]

For voice applications specifically (e.g., IP telephony)—the phones are IP-attached Ethernet devices, and there are many potential benefits:

 Use of a single wiring closet connection for both phone and PC  Choices in the application of power to the phone  Assignment of IP addresses without wholesale renumbering  Ease of adds, moves, and changes  Guarantee of voice quality, even under congested conditions  Overall redundancy and reliability of the network

You can maintain separate logical networks for your voice and data networks, even though the physical infrastructure is physically the same.

IP phones use DHCP to obtain an IP address, then locates the Call Manager (CM) on the “phone network” then it FTP’s its configuration and is ready to use.

The IP Phones themselves are “programmable” via XML (i.e., you can create links from the LCD screen on the phone to whatever you define). There is a white paper on it from Cisco at http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/devguide/index.htm

Packetized voice traffic is routed from source end station to destination end station using layer 2 and layer 3 protocols. Either static tables pre-programmed in each switch or a dynamic routing protocol implemented throughout the network can be used to determine the route.

[SLIDE 6—QoS issues]

QoS

VoIP and Delay

Routers and specifically IP networks offer some unique challenges in controlling delay and delay variation. Traditionally, IP traffic has been treated as "best effort," meaning that incoming IP traffic is allowed to be transmitted on a first-come, first-served basis. Packets have been variable in nature, allowing large file transfers to take advantage of the efficiency associated with larger packet sizes. These characteristics have contributed to large delays and large delay variations in packet delivery. However, recent efforts have been made through standards to support traffic that is more sensitive to delay and delay variation.

[SLIDE 7—QoS issues at the edge]

The second part of supporting delay-sensitive voice traffic is to provide a means of prioritizing the traffic within the router network. RFC 1717 breaks down large packets into smaller packets at the link layer. This reduces the problem of queuing delay and delay variation by limiting the amount of time a voice packet must wait in order to gain access to the trunk.

Weighted Fair Queuing or priority queuing allows the network to put different traffic types into specific QoS queues. This is designed to prioritize the transmittal of voice traffic over data traffic. This reduces the potential of queuing delay. The Catalyst 6509 switches support QoS features such as packet classification and marking, scheduling, policing, and congestion avoidance. Catalyst modules provide extensive per-port queuing to guarantee that voice traffic is given highest priority. Dedicated voice queues can be configured such that QoS is maintained end to end through the switch and across the network. Network resilience features such as Layer 2 and Layer 3 load balancing, redundant system elements, and fast fail-over mechanisms maintain highest levels of system availability. QoS policies are enforced using Layer 2, 3, and 4 information such as 802.1p, IP Precedence, and Layer 4 port numbers. Catalyst 6509 switches can have multiple queues with configurable thresholds via Weighted Random Early Detection (WRED), Weighted Round Robin (WRR), and type-of-service/class-of-service (ToS/CoS) mapping mechanisms to ensure that QoS is maintained as packets traverse the network. Resource Reservation Protocol (RSVP) priority mapping can also be used, ensuring timely delivery of time-sensitive intranet applications.

Jitter is another factor that affects delay. Jitter occurs when there is a variation between when a voice packet is expected to be received and when it actually is received, causing a discontinuity in the real- time voice stream. Voice devices such as the Cisco 3600 router (we’re using a 3640) compensate for jitter by setting up a playout buffer to playback voice in a smooth fashion. Playout control is handled through RTP encapsulation, either by selecting adaptive or non-adaptive playout-delay mode.

QoS tools (to deal with packet loss, jitter, and delay) – see p.190 in VoIP book:

At the edge of the network:

 Additional bandwidth  Compressed real-time transport protocol--  Queuing (Weighted fair queuing, Custom queuing, Priority queuing, Class-based WFQ)  Packet classification (IP Precedence, policy routing, Resource reservation protocol, IP Real-time transport protocol Reserve, IP RTP Priority)  Shaping traffic flows and policing (Generic traffic shaping, Frame relay traffic shaping, Committed access rate)  Fragmentation (Multi-class multilink point to point protocol, frame relay forum, Maximum transmission unit, IP MTU)

[SLIDE 8—QoS issues in the backbone]

In the backbone of the network:

 Packet over SONET  IP and ATM inter-working  WRED—weighted random early drop/detect  DWFQ—distributed weighted fair queuing [SLIDE 9—powering the phones]

Power options

A single copper wire connection can support both the telephone set and the desktop PC Ethernet connection at 10 or 100 Mbps. This configuration is possible due to integrated 10/100 switch ports on a Cisco IP phone.

On the port of the phone that connects to the IP network switch, 48 volts of power can be delivered in two ways: 1) The first method utilizes new inline power 10/100 Ethernet switch ports on some Cisco Catalyst switches that automatically detect the presence of the phone and apply power on the same four wires that carry the Ethernet signals. 2) The second method works with any existing switching platform and uses a Catalyst switch-powered patch panel that inserts power on the other four wires of the eight-wire bundle to the desktop. 3) A third option to use an AC power adapter and plug the phone into a wall socket.

The method chosen to power the phone set is a function of the amount of copper wire to the desktop, the type of switching equipment installed in the wiring closet, and the overall power backup strategy of the corporation. There are obvious centralized power control issues to consider here which could directly affect network availability. We are currently testing a small switch that provides power to the IP phones over the existing UTP cabling. We are also testing the wall power option. As part of our future VoIP test procedure, we will be testing some cards for 6509 switches that will deliver the power over the network cable.

The Cisco phone discovery feature automatically detects the presence of an IP phone and supplies inline power. This means that network administrators can maintain centralized control without the need to manually enable each port to supply inline power. The phone discovery mechanism is intelligent enough to differentiate between an IP phone and a network interface card, and will not supply inline power to a network interface card or other device not designed to use inline power.

To support the new demand for phone power provided with the inline power feature, Cisco has developed a new 2500-watt power supply for the Catalyst 6000 family. This power supply has been designed to work in Catalyst 6000 family chassis' that will be loaded with inline power line cards and IP phones. For fault tolerance, two power supplies can be deployed in a single chassis to guard against a single power supply failure.

[SLIDE 10—the NETS VoIP team]

NETS VoIP team and components of our test:

Work from slide 10—team/resources

[SLIDE 11—the sandbox]

Work from slide 11—the sandbox

[SLIDE 12—project goals]

Work from slide 12—project goals [SLIDE 13—wrap up—related pages and questions?]

Work from slide 13—related materials and questions (indicate that page will be updated over time and is presently out of date pending me updating it)

#START of offline items

NCAB VoIP Presentation – 17-Jan-2001, 10am-12pm, ML Chapman Room

“VoIP offline items”

Equipment ordered (Most pieces of the current “VoIP puzzle” have arrived):  Cisco “Media Convergence Server” 7835 (a Compaq Proliant DL380 (specs available offline if anybody is interested) with Windows 2000 server) running the Cisco Call Manager software  Cisco 3640 router with E/M lines to connect to existing PBX and provide calls outside VoIP test network  25 IP Phones (Cisco model 7960) ordered (can be powered over network cable or with power cord)  right now, we have a loaner Cisco 3524 in-line power switch which powers the phones over the network cable; also have some test phone deployed using wall power. [we also have powered cards for a 6509s, but are not deploying/testing them yet—we will]

Current test bed setup:  have a VoIP “sandbox” set up where we are now learning how the pieces fit together, and are going over various configuration issues (just done over the last couple of days).

Ongoing/Future testing plans:  Test various configuration components (e.g., interaction with existing PBX features, phone configuration options, user configuration options, etc.)  We want to enable QoS features in the VoIP network and test effect with and without these in place  Are considering ordering a Unified Messaging server to test allow us to handle voice mail, email, and test interaction with our existing PBX phone mail and the Cisco Call Manager.  Also want to test the 6509 cards that provide in-line power over the network cable.

Check the “VoIP project” page off the NETS main web page under “Projects” for updates and more information [URL=http://www.scd.ucar.edu/nets/projects/voip/]

***** Other VoIP “overview details”:

VoIP signaling has three distinct areas: 1) Signaling from the PBX to the router 2) Signaling between routers 3) And signaling from the router to the PBX.

The corporate intranet appears as a trunk line to the PBX, which will signal the corporate intranet to seize a trunk. Signaling from the PBX to the intranet may be any of the common signaling methods used to seize a trunk line, such as FXS or E&M signaling. The PBX then forwards the dialed digits to the router in the same manner the digits would be forwarded to a Telco switch. Within the router the “Dial Plan Mapper” maps the dialed digits to an IP address and signals a Q.931 Call Establishment Request to the remote peer that is indicated by the IP address. Meanwhile, the control channel is used to set up the Real Time Protocol (RTP) audio streams, and the RSVP protocol is used to request a guaranteed quality of service.

When the remote router receives the Q.931 call request it signals a line seizure to the PBX. After the PBX acknowledges, the router forwards the dialed digits to the PBX, and signals a call acknowledgment to the originating router.

In connectionless network architectures like IP, the responsibility for session establishment and signaling resides in the end stations. To successfully emulate voice services across an IP network, enhancements to the signaling stacks are required. For example, an H.323 agent is added to the router for standards-based support of the audio and signaling streams. The Q.931 protocol is used for call establishment and tear down between H.323 agents or end stations. RTCP, the Real Time Control Protocol, is used to establish the audio channels themselves. A reliable session-oriented protocol, TCP, is deployed between end stations to carry the signaling channels. RTP, the Real Time Transport Protocol, which is built on top of UDP, is used for transport of the real-time audio stream. RTP uses UDP as a transport mechanism because it has lower delay than TCP, and because actual voice traffic, unlike data traffic or signaling, tolerates low levels of loss and cannot effectively exploit retransmission. Table 1 depicts the relationship between the ISO reference model and the protocols used in IP voice agents

Table 1 ISO Reference Model and H.323 Standards ISO Protocol Layer ITU H.323 Standard

Presentation G.711,G.729, G.729a, etc.

Session H.323, H.245, H.225, RTCP

Transport RTP,UDP

Network IP, RSVP, WFQ

Link RFC1717(PPP/ML), Frame, ATM, etc.

Standards

 G.711---Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct format for digital voice delivery in the PSTN or through PBXs.  G.729---Describes CELP compression where voice is coded into 8-kbps streams. There are two variations of this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both provide speech quality similar to 32-kbps ADPCM.  See p.173 of VoIP book for more voice coding standards

*****

VoIP Signaling VoIP signaling has three distinct areas: signaling from the PBX to the router, signaling between routers, and signaling from the router to the PBX. The corporate intranet appears as a trunk line to the PBX, which will signal the corporate intranet to seize a trunk. Signaling from the PBX to the intranet may be any of the common signaling methods used to seize a trunk line, such as FXS or E&M signaling. In the future, digital signaling such as CCS or QSIG will become available. The PBX then forwards the dialed digits to the router in the same manner the digits would be forwarded to a Telco switch.

VoIP Addressing

In an existing corporate intranet, an IP addressing plan will be in place. To the IP numbering scheme, the voice interfaces will appear as additional IP hosts, either as an extension of the existing scheme, or with new IP addresses.

Translation of dial digits from the PBX to an IP host address is performed by the dial plan mapper. The destination telephone number, or some portion of the number, will be mapped to the destination IP address. When the number is received from the PBX, the router compares the number to those mapped in the router table. If a match is found, the call is routed to the IP host. After the connection is established, the corporate intranet connection is transparent to the subscriber.

VoIP Routing

One of the strengths of IP is the maturity and sophistication of its routing protocols. A modern routing protocol, such as EIGRP, is able to take delay into consideration in calculating the best path. These are also fast converging routing protocols, which allows voice traffic to take advantage of the self-healing capabilities of IP networks. Advanced features, such as policy routing and access lists, make it possible to create highly sophisticated and secure routing schemes for voice traffic.

RSVP can be automatically invoked by Cisco's VoIP gateways to insure that voice traffic is able to use the best path through the network. This can include segments of arbitrary media, such as switched LANs or ATM networks. Some of the most interesting developments in IP routing are the development of Tag Switching and other IP switching disciplines. Tag Switching provides a way of extending IP routing, policy, and RSVP functionality over ATM and other high-speed transports. Another benefit of Tag Switching is its traffic engineering capabilities, which are needed for the efficient use of network resources. Traffic engineering can be used to shift traffic load based on different predicates, such as time of day.

The next section explores developments in IP that will control network delay and delay variation.

VoIP and Delay

Routers and specifically IP networks offer some unique challenges in controlling delay and delay variation. Traditionally, IP traffic has been treated as "best effort," meaning that incoming IP traffic is allowed to be transmitted on a first-come, first-served basis. Packets have been variable in nature, allowing large file transfers to take advantage of the efficiency associated with larger packet sizes. These characteristics have contributed to large delays and large delay variations in packet delivery. However, recent efforts have been made through standards and Cisco's own unique efforts to support traffic that is more sensitive to delay and delay variation. Resource Reservation Protocol or RSVP allows us to reserve resources in the network by the end station. This allows us to allocate queues for different types of traffic, helping us to reduce delay and delay variation inherent in current IP networks.

The second part of supporting delay-sensitive voice traffic is to provide a means of prioritizing the traffic within the router network. RFC 1717 breaks down large packets into smaller packets at the link layer. This reduces the problem of queuing delay and delay variation by limiting the amount of time a voice packet must wait in order to gain access to the trunk.

Weighted Fair Queuing or priority queuing allows the network to put different traffic types into specific QoS queues. This is designed to prioritize the transmittal of voice traffic over data traffic. This reduces the potential of queuing delay.

These next points are raised to explain some of the developments taking place within the IP community. One of the issues with regard to the deployment of IP was the different underlying Layer 2 protocols and the need to provide some means of address resolution. Address resolution can be statically defined within a table, employ some form of broadcast or use a central address resolution server.

Another development within the IP community, explained earlier, is the use of DHCP and DNS to provide a level of abstraction. DHCP allows you to ignore your IP address; whereas, DNS allows you to ignore the address of the person or thing you wish to contact. Perhaps similar mechanisms will help evolve the telephone from a physical to a logical device.

Finally, one of the "hot" topics is the use of an enterprise directory service to define identities and policies. This is a level of abstraction beyond IP addressing and it will be interesting how this could be potentially tied to voice communications.

#END of offline items

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