Cisco_technical Voip_white Paper 09-29-2005

White Paper

Technical Basics of VoIP

Introduction The acronym VoIP stands for voice over Internet Protocol, but it has taken on more meanings than just using a particular protocol to transmit voice. It can mean a desktop device or IP phone, it can mean making phone calls over the public Internet, and it can refer to the transport of traditional voice calls between two time-division multiplexing (TDM) switches using packetization and IP. Some of the technologies and applications with which VoIP is used are detailed in this paper. To understand more about VoIP, it might help to see where some common non-VoIP elements fit in: Primary Rate Interface (PRI), channel associated signaling (CAS), and Signaling System 7 (SS7).

PRI/CAS over T1 PRI and CAS are two traditional methods for specifying the signaling arrangement over a T1 digital interface between two voice switches. The signaling passes over a single D channel, as in PRI, or the signals travel over individual voice channels with the voice call on that channel. A common use of this type of circuit is between a TDM private branch exchange (TDM PBX) and a TDM Class 5 switch. In VoIP networks, PRI and CAS interfaces are commonly used at the network edges. They terminate on gateway ports that take in the signal traffic in TDM format, change the signaling and the bearer traffic to an IP packet format, and transport it out an IP port such as Ethernet. On a customer premises, a PRI or CAS T1 interface is used as an interface to the TDM key system or the PBX. It can also be used to receive or send traffic from a Class 5 switch connected to the public switched telephone network (PSTN), as shown in Figure 1. When the signaling is received as PRI or CAS at the router or gateway T1 port, the signaling traffic is converted to a VoIP signaling packet protocol like Media Gateway Control Protocol (MGCP) or H.248. The signaling packet protocol is then transported to the softswitch over an IP infrastructure.

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Figure 1 PRI/CAS Interface from TDM to PSTN VoIP Network

One advantage of converting everything to IP at the edge is then being able to carry all types of internal traffic and applications over IP, sharing the infrastructure. This distributed architecture avoids having to backhaul a PRI TDM circuit all the way back to the location of the softswitch. Some softswitches have PRI interface cards on the same hardware platform as the call control, and backhaul is required.

SS7 The huge growth in cellular, mobile, and Short Message Service (SMS/texting) traffic has resulted in increases in SS7 signaling traffic. Signaling System 7 has been a backbone component of the telephone network for decades. It already uses packet technology to transport signaling information for phone calls over a signaling network that is separate from the route that the voice conversation traverses. This great growth in traffic placed a burden on the traditional SS7 network, where the largest channel was capable of 56 Kbps of bandwidth. Several vendors have made SS7 signaling packets compatible with IP networks, allowing for SS7 signaling traffic to traverse networks that have 10 Mbps, 100 Mbps, and Gigabit Ethernet connections. This is done by converting the SS7 signaling to SS7-over-IP protocols SCTP (RFC2960), M2PA, M2UA, M3UA, and SUA. Figure 2 shows where these new protocol stacks line up relevant to the traditional SS7 protocol design.

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Figure 2 SS7 Protocol Architectures

An SS7 signaling gateway controller functions similarly, converting Message Transfer Part Layer 1 (MTP1), MPT2, MTP3, and Skinny Client Control Protocol (SCCP) messages into IETF SIGTRAN messages (SS7 over IP). Learn more about SIGTRAN at http://www.ietf.org/html.charters/sigtran-charter.html. The transformation from TDM SS7 to SS7oIP has resulted in numerous benefits. Now SS7 signaling traffic can be carried over higherbandwidth connections and be transported with other data traffic, lowering operational costs and increasing application flexibility. Existing IP packet backbone networks can run at higher utilization levels as well. The main reason for the large bandwidth savings is that SS7oIP systems do not need Fill In Signal Units (FISUs) added to the Message Signal Units (MSUs) on the IP side (see Figure 3). The FISUs are terminated on the TDM side of the signaling gateway to yield bandwidth savings.

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Figure 3 Bandwidth Efficiencies of SS7oIP Compared to SS7/TDM

To implement SS7oIP, three logical components are typically needed: • A signaling gateway controller performing the function of the Signal Transfer Point (STP) (in some products it is embedded in the IP softswitch and in others it is an optional separate signaling controller) • An SS7 gateway that can provide traditional A links and F links on the TDM side and convert to IP/SIGTRAN on the IP side • Trunking gateways to convert Intermachine Trunk bearer TDM traffic to packetized IP traffic

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Figure 4 shows a representative diagram of SS7oIP with interfaces to the PSTN. Figure 4 PSTN Voice Transit with SS7oIP Integrated Voice and Signaling

There are methods within IP networks to implement quality-of-service (QoS) capabilities in SS7oIP, which helps ensure that SS7 traffic is differentiated and given precedence over other types of traffic sharing the network bandwidth. QoS also allows for the various types of SS7oIP messages to be selectively differentiated.

IP Telephony for SMBs and Enterprises IP telephony is about making phone calls and running other applications over the same infrastructure simultaneously. IP telephony also includes video telephony, information services at the phone, and user control of features. This section focuses on solutions for enterprises and small and medium-sized businesses (SMBs). Consumer IP telephony is covered in a separate section. IP telephony is a large subset of what the industry addresses with the term VoIP. IP telephony generally refers to the customer premises equipment (CPE), the software, and the methods for using this equipment to make phone calls originating from a LAN infrastructure. Following are six examples of these elements.

IP PBX An IP PBX does what a TDM PBX can do and more. It is usually integrated with IP telephones to offer special features. The protocol between the IP PBX and the IP phone can be standards-based or proprietary. Some IP phones and many IP PBXs are capable of running multiple protocols. PBXs are typically owned and operated by an enterprise or small business, but they may also be owned and managed by the service provider or local exchange carrier (LEC). The nature of IP communications allows for the IP PBX to reside off premises. The IP PBX is different from TDM PBXs in many ways. The lines to the IP PBX do not have to be physically connected to the PBX (also called the call agent). The call agent is often software running on a server. The server gains connectivity for communication to the phones through the Cisco Systems, Inc. All contents are Copyright © 1992–2005 Cisco Systems, Inc. All rights reserved. Important Notices and Privacy Statement. Page 5 of 16

LAN switching infrastructure. If the PBX is located remotely, the WAN may be used to connect the call agent with remote phones. It is not uncommon to have an abbreviated dial plan supported across quite a large WAN network. Mobility is another advantage of IP PBXs over traditional TDM PBXs. People can go to a branch office, sign into a phone, and have calls to their local number terminate on that phone set. The mobility comes from the fact that an IP phone is just an endpoint associated with a user. Users can move their phones, push their features to another phone, or use softphones to move their features to their PCs.

IP Phone IP phones can derive their power from Power over Ethernet (PoE) ports on the LAN switch or use AC adapters. They can also have the traditional keypad buttons and multiple lines. A few of the differences are the use of softkeys and displays. The softkeys can be programmed to support applications not traditionally found on telephones. Using Extensible Markup Language (XML), the display can become a terminal for executing applications. For example, a corporate directory database can be displayed right on the phone. It can be used for a food menu or list of services in a hotel. These capabilities have provided new applications for industries such as healthcare, education, finance, and hospitality.

IP Key System An IP Key system works like a TDM Key system. The feature set is usually a subset of what is available on an IP PBX, and usually the price is lower. An IP Key system is a software stack running on a CPE router that locally manages the call control of the phone, which can be IP phones or analog phones with an adapter. This allows providers to offer multiple services on one CPE device, such as telephony, Internet access, security, and long-distance access.

Analog Telephony Adapter An analog telephony adapter (ATA) is used in IP telephony environments to allow analog phones and fax machines to communicate over an IP network using RJ-11 jacks. ATAs are also used with data modems and point-of-sale credit-card transaction terminals. The ATA usually has an Ethernet port for connection to the LAN or router on the customer premises. The ATA converts the analog signals to IP packets for transport over the IP switching and routing network.

Softphone A softphone is software that puts IP phone capability on a device other than an IP phone, for example a PC. The PC emulates the functions of an IP phone for making phone calls, typically using a headset with a microphone to create a hands-free environment. This configuration can also be used in environments like call centers. The softphone is also used by traveling or remote workers who want to have their local number ring wherever they are connected to the Internet, at home or in their hotel, without having to carry their IP phone with them.

LAN Switch The ability of IP telephony systems to work in a business setting is highly dependent on the LAN switching environment. IP phones can derive their power from PoE ports on the LAN switch; 10 Mbps, 100 Mbps, and Gigabit Ethernet ports are available as uplinks to the LAN switch. A phone call may not require this much bandwidth, but other applications that run on the phone or on PCs attached to the phone may.

Media and Signaling Gateways Media gateways occur in enterprise networks and are also deployed in the core of service provider IP networks. (Service providers typically require larger gateway capacity to aggregate traffic from many IP sources—enterprise, SMB, and residential). The function of a gateway is to convert media from one format to another and move it from one network to another. A gateway can take a voice conversation that originates Cisco Systems, Inc. All contents are Copyright © 1992–2005 Cisco Systems, Inc. All rights reserved. Important Notices and Privacy Statement. Page 6 of 16

from the PSTN on a TDM circuit and convert the signals to an IP protocol. Another type of gateway is an IP-to-IP gateway, also called a session border controller (SBC). The IP-to-IP gateway can provide a demarcation point between IP networks and protect private addressing on each side of the network for security purposes. It can also convert between two different VoIP protocols or normalize features between a common protocol on both sides. SBCs may also perform Network Address Translation (NAT) traversal functions. SBCs are becoming more prevalent as more networks interconnect natively with IP. A “Session Border Controller” (SBC) is a device that generally sits between two administrative domains (for example, between two service providers) and is responsible for routing VoIP calls between the two domains. An SBC may route only call signaling or it may route both the call signaling and the media between the administrative domains. Most will route both. The SBC may also serve as a protocol interworking function to interwork between various VoIP protocols, including H.323 and SIP (from Techabulary.com). NAT traversal is also an important function that many SBCs provide. An alternative to SBCs for NAT traversal is the use of a STUN server.

Bandwidth Consumption for VoIP All traffic is packetized in an IP environment, whether it is signaling or bearer (actual media) traffic. The size of the packets for voice traffic depends on the packetization scheme or algorithm. In addition to the actual payload of the packet, there is header information that identifies where the packet came from, where it is going, how big it is, and whether it is a part of a group of packets. Packets can be compressed according to various standards; for an overview and bandwidth consumption numbers for some of the most common compression techniques, visit: http://www.cisco.com/warp/public/788/pkt-voice-general/bwidth_consume.html. The compression technique chosen can depend on how much bandwidth is available in the network. It is not uncommon to use one method for the intra-LAN telephony traffic, where bandwidth is relatively abundant, and another for sending traffic over the WAN, where bandwidth capacity is more expensive.

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Figure 5 highlights the conversion of a packet from analog TDM to digital for a VoIP call. Figure 5 Packet Conversion for VoIP

Using different codecs will yield different results depending on the codec chosen, the network, the application, and the user’s behavior and perception. Table 1 is a reference for using different codecs for different applications without any other consideration with grades assigned to the relative strength of the codec for the application. Table 1.

Voice Codecs

Encoding Compression

Mean Opinion Score

Native Bit Rate Kbps

Voice Quality

BW

DTMF

Dual Comp

CPU

Music on Hold

G.711 PCM

4.1

64

A

D

A

A

A

A

G.726 ADPCM

3.85

32

B

C

B

B

B

B

G.728 LD-CELP

3.61

16

C

B

B

C

C

C

G.729 CS-ACELP

3.92

8

A

A

B

B

C

C

G.729a CS-ACELP

3.7

8

B

A

C

C

B

D

G.723.1 ACELP

3.65

5.3

C

A

C

D

C

D

Quality of Service QoS is essential to provide high-quality voice calls over IP. Adding bandwidth helps, but poor-quality voice is not normally caused by lack of bandwidth but rather by contention for queues. One important factor for high-quality voice calls is to be able to identify and select voice traffic packets to be treated with higher priority than other nonvoice packets in the network, such as e-mail, an example of a best-effort application. There are VoIP networks that are entirely best-effort with no prioritization for the VoIP packets. Those networks do not have consistently good voice quality. Some general guidelines for QoS are shown in Figure 6.

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Figure 6 QoS Requirements for Voice

Delay of voice traffic through the network can have a negative effect on the conversation. Therefore, it is critical to minimize latency and jitter (delay variation) for VoIP traffic so that the result is a smooth flowing conversation and a smooth flow of packets. Figure 7 shows some of the causes of these attributes. Figure 7 Elements that Affect Latency and Jitter

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Figure 8 outlines the considerations taken when planning VoIP QoS in a campus environment. Figure 8 QoS Considerations in Avoiding Loss, Latency, and Jitter

Hosted IP Telephony Versus Managed IP Telephony Two services that providers are offering with VoIP telephones are hosted and managed IP telephony. In hosted service, the call switching agent does not reside on the customer premises. This is similar to traditional TDM Centrex, but in many cases the term Centrex is avoided to differentiate service and tariff implications. The features offered with a hosted IP telephony service often exceed those from traditional Centrex. The business arrangement is typically similar, where the provider leases the phones and the feature sets are packaged in the offering. A difference between traditional Centrex and hosted IP telephony is the local loop. With traditional Centrex, twisted pairs typically run from the customer premises back to the central office. Hosted IP telephony service is provided over a broadband connection often bundled with Internet access, long distance, and voice mail to create a local integrated services package. Figure 9 depicts a broadband solution to a small business office using hosted IP telephony service architecture.

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Figure 9 Hosted IP Telephony over Broadband

Managed IP telephony is a service where the service provider is responsible for managing the VoIP telephony equipment. It can be hosted or on the customer premises and based on IP PBX or centralized on a feature server. The service provider manages dial plan creation, configuration of the equipment, administration functions, and sometimes moves, adds, and changes. The service provider’s network infrastructure is the key to providing this type of service. VPNs, QoS, service-level agreements (SLAs), and backup and recovery are often included in the contract. These can lead to other managed offerings like storage area networking and managed security. Figure 10 shows an example of managed VoIP service architecture. Cisco Systems, Inc. All contents are Copyright © 1992–2005 Cisco Systems, Inc. All rights reserved. Important Notices and Privacy Statement. Page 11 of 16

Figure 10 Managed Voice Services

Consumer VoIP Services Consumer VoIP services are being offered today using various methods. In some instances, the provider owns and manages the local loop and the aggregation of those loops. In other offerings, the VoIP service rides over some other carrier’s broadband service and is delivered on a besteffort basis. The difference between a managed loop and a best-effort loop can make the difference between good and bad voice quality. The trade-off to the consumer is usually the price. If a service is offered at US$15 a month and provides good quality service most of the time, the market has shown that to be attractive. In fact it may be an improvement over typical cellular phone quality.

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Cable operators are offering VoIP service with QoS built into the architecture. The bundle usually includes television, telephony, and broadband Internet access, otherwise known as “triple play,” offered over one coaxial cable to the home. The standard they follow is known as DOCSIS. The aggregation point, known as a cable modem termination server (CMTS), is where policy is implemented to ensure that VoIP traffic (and quality) is not compromised by other traffic on the network (Figure 11). Figure 11 “Triple Play” Service Delivered by PacketCable™ Architecture

VoIP as Best-Effort Service The consumer can also have best-effort VoIP service over cable, DSL, or other broadband connection. An example is to use cable service for Internet access and have an application service provider ride over that cable plant without any guarantees, as opposed to having the cable company implement VoIP over DOCSIS. In this situation, the VoIP traffic is truly contending with everything on the neighborhood cable plant network, including other traffic from the originating subscriber. See Figure 12.

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Figure 12 Residential VoIP Best-Effort Offering

GR303 Many residential telephony deployments today use GR303 architecture as a method of oversubscribing the central office’s switch port out to the local loops. There are several directions for the LEC to take with regard to the remote terminal plant that is deployed in the network that runs GR303. Some IP softswitches offer native interfaces for GR303. Running GR303 links back to the central office to connect directly into the softswitch is an option in those instances. Another option is to terminate the GR303 links (still preserving the remote terminals) on a “reverse GR303 gateway” and then run an IP signaling protocol into the softswitch (Figure 13). Figure 13 GR303 Reverse Gateway

A third option is to deploy next-generation or broadband data link connections (DLCs), which come equipped with IP stacks on the uplink for communicating with the softswitch. Cisco Systems, Inc. All contents are Copyright © 1992–2005 Cisco Systems, Inc. All rights reserved. Important Notices and Privacy Statement. Page 14 of 16

Summary Voice over IP has many definitions. The packetization of voice traffic is now commonplace using codecs and DSP technology. The development of IP-based applications using VoIP has created new opportunities for service providers where voice becomes one of many applications running over the network. The capabilities of the equipment at the customer premises, the last mile, the edge, and the core of the network determine what types of services can be offered. Some type of broadband in the local loop in the form of fiber, DSL, T1, or highfrequency cable is usually required. Another common requirement for any VoIP deployment is QoS in the core and access networks, which is the most technically challenging aspect of deploying VoIP. Creating a high-quality VoIP experience that rivals the PSTN requires a sound architecture and implementation of QoS, including the ability to differentiate traffic types and treat them appropriately.

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