Cisco Voice over IP (CVoice) (Authorized Self-Study Guide) (2nd Edition)

Configuring Cisco Systems voice equipment to interface with other equipment requires an understanding of the signaling that conveys supervision between the systems. Proper troubleshooting also requires an understanding of these signaling systems.

This section describes the various signaling systems used between telephony systems, such as common channel signaling and channel associated signaling. It also explores signaling between PBXs, signaling between PBXs and COs, and specialized signaling, such as ISDN.

Channel Associated Signaling

Channel associated signaling (CAS) is a signaling method commonly used between PBXs. Although this can manifest itself in many forms, some methods are more common than others. Signaling systems can also be implemented between a PBX and a Cisco voice device.

T1 Channel Associated Signaling

PBXs and Cisco devices use T1 and E1 interfaces to convey voice. Originally, this was the main purpose of T1, which carries signaling information using two methodologies: CAS and common channel signaling (CCS). Figure 2-31 illustrates the format of the T1 digital signal.

The characteristics of the T1 digital signal format are as follows:

• A T1 frame is 193 bits long, 8 bits from each of the 24 time slots (digital service zeros [DS0s]), plus 1 bit for framing. A T1 repeats every 125 microseconds, resulting in 8000 samples per second (8 bits * 24 time slots + 1 framing bit * 8000 samples per second = 1.544 Mbps).

• T1 has two major framing and format standards:

- Super Frame (SF), or D4, specifies 12 frames in sequence. The D4 framing pattern used in the F position in Figure 2-31 is 100011011100 (a 1 goes with the first frame, a 0 goes with the second frame, a 0 goes with the third frame, and so on, all the way through 12 frames). This unique framing pattern allows the receiving T1 equipment to synchronize within four frames, since any four consecutive frame bits are unique within the 12-bit pattern. Because there are 8000 T1 frames transmitted per second, 8000 F bits are produced and used for framing.

- Extended Superframe (ESF) format was developed as an upgrade to SF and is now dominant in public and private networks. Both types of formatting retain the basic frame structure of one framing bit followed by 192 data bits. However, ESF repurposes the use of the F bit. In ESF, of the total 8000 F bits used in T1, 2000 are used for framing, 2000 are used for cyclic redundancy check (CRC) (for error checking only), and 4000 are used as an intelligent supervisory channel to control functions end to end (such as loopback and error reporting).

Because each DS0 channel carries 64 kbps, and G.711 is 64 kbps, there is no room to carry signaling. Implemented for voice, the T1 uses every sixth frame to convey signaling information. In every sixth frame, the least significant bit (LSB) for each of the voice channels is used to convey the signaling, as shown in Figure 2-32. Although this implementation detracts from the overall voice quality (because only 7 bits represent a sample for that frame), the impact is not significant. This method is called robbed-bit signaling (RBS). When SF employs this method, the signaling bits are conveyed in both the 6th (called the "A" bit) and 12th (called the "B" bit) frames. For control signaling, A and B bits provide both near- and far-end off-hook indication.

The A and B bits can represent different signaling states or control features (on hook or off hook, idle, busy, ringing, and addressing). The robbed bit is the least significant bit from an 8-bit word.

ESF also uses RBS in frames 6, 12, 18, and 24 to yield four signaling bits, providing additional control and signaling information. These four bits are known as the A, B, C, and D bits.

Because the signaling occurs within each DS0, it is referred to as in band. Also, because the use of these bits is exclusively reserved for signaling each respective voice channel, it is referred to as CAS.

The robbed bits, depicted in Figure 2-33, are used to convey E&M status or FXS/FXO status and provide call supervision for both on hook and off hook.

T1 CAS has the following characteristics:

• SF has a 12-frame structure and provides AB bits for signaling.

• ESF has a 24-frame structure and provides ABCD bits for signaling.

• DTMF, or tone, can be carried in band in the audio path. However, other supervisory signals must still be carried via CAS.

E1 Channel Associated Signaling

In E1 framing and signaling, 30 of the 32 available channels, or time slots, are used for voice and data. Framing information uses time slot 1, while time slot 17 (E0 16) is used for signaling by all the other time slots. This signaling format, illustrated in Figure 2-34, is also known as CAS because the use of the bits in the 17th time slot is exclusively reserved for the purpose of signaling each respective channel. However, this implementation of CAS is considered out of band, because the signaling bits are not carried within the context of each respective voice channel, as is the case with T1. E1 CAS is directly compatible with T1 CAS, because both methods use AB or ABCD bit signaling. Although the signaling for E1 CAS is carried in a single common time slot, it is still referred to as CAS because each individual signaling time slot represents a specific pair of voice channels.

In the E1 frame format, 32 time slots make up a frame. A multiframe consists of 16 E1 frames, as depicted in Figure 2-35.

The time slots are numbered 1 though 32, although the channels are numbered 0 through 31, as shown in Figure 2-35. Multiframe time slots are configured as follows:

• Time slot 1 carries only framing information.

• Time slot 17, in the first frame of the 16-frame multiframe, declares the beginning of the multiframe, which is indicated by the M symbol in Figure 2-35.

• The remaining slot 17s carry signaling information for all the other time slots:

  - Slot 17 of the first frame declares the beginning of a 16-frame multiframe (M).

  - Slot 17 of the second frame carries ABCD for voice slot 2 (X) and ABCD for voice slot 18 (Y).

  - Slot 17 of the third frame carries ABCD for voice slot 3 (X) and ABCD for voice slot 19 (Y).

  This process continues for all the remaining frames.

Common Channel Signaling Systems

Common channel signaling (CCS) differs from CAS in that all channels use a common channel and protocol for call setup. Using E1 as an example, a signaling protocol, such as ISDN Q.931, would be deployed in time slot 17 to exchange call-setup messages with its attached telephony equipment, as seen in Figure 2-36.

Examples of CCS signaling are as follows:

• Proprietary implementations Some PBX vendors choose to use CCS for T1 and E1 and implement a proprietary CCS protocol between their PBXs. In this implementation, Cisco devices are configured for Transparent Common Channel Signaling (T-CCS) because the Cisco devices do not understand proprietary signaling information.

• Integrated Services Digital Network (ISDN) ISDN uses Q.931 in a common channel to signal all other channels.

• Q Signaling (QSIG) Like ISDN, QSIG uses a common channel to signal all other channels.

• Digital Private Network Signaling System (DPNSS) DPNSS is an open standard developed by British Telecom for implementation by any vendor who chooses to use it. DPNSS also uses a common channel to signal all other channels.

• Signaling System 7 (SS7) SS7 is an out-of-band network implemented and maintained by various telephone companies and used for signaling and other supplemental services.

The following discussions elaborate on various CCS implementations. Note that proprietary implementations are not discussed because they vary widely among vendors.

ISDN

ISDN (Integrated Services Digital Network) is an access specification to a network. You may have studied ISDN as an access method for dial-up data systems. Because it is a digital system, ISDN makes connections rapidly.

ISDN can be implemented in two different ways: BRI (Basic Rate Interface) and PRI (Primary Rate Interface). BRI features two bearer (B) channels, while PRI supports 23 (for T1) or 30 (for E1) B channels. Each implementation also supports a data (D) channel, used to carry signaling information (CCS).

The following are benefits of using ISDN to transmit voice:

• Each B channel is 64 kbps, making it perfect for G.711 PCM.

• ISDN has a built-in call control protocol known as ITU-T Q.931.

• ISDN can convey standards-based voice features, such as call forwarding.

• ISDN supports standards-based enhanced dial-up capabilities, such as Group 4 fax and audio channels.

Note

ISDN BRI voice is commonly used in Europe, while ISDN PRI voice is used worldwide.

Figure 2-37 shows the architecture of an ISDN network. The B channel carries information, such as voice, data, and video, at 64 kbps. The D channel carries call signaling between customer premises equipment (CPE) and the network, usually as the Q.931 protocol but sometimes as the QSIG protocol.

BRI operates using the average local copper pair. It uses two B channels and one signaling channel, which is written as 2 B + D.

PRI implemented on T1 uses 23 B channels and one signaling channel, which is written as 23 B + D. PRI implemented on E1 uses 30 B channels and one signaling channel, which is represented as 30 B + D.

ISDN's Q.931 protocol, which operates at Layer 3 of the OSI (Open System Interconnection) model, uses a standard set of messages to communicate, as illustrated in Figure 2-38.

These standard messages cover the following areas:

• Call establishment Initially sets up a call. Messages travel between the user and the network. Call establishment events include alerting, call proceeding, connect, connect acknowledgment, progress, setup, and setup acknowledgment.

• Call information phase Data sent between the user and the network after the call is established. This allows the user to, for example, suspend and then resume a call. Events in the call information phase include hold, hold acknowledgment, hold reject, resume, resume acknowledgment, resume reject, retrieve, retrieve acknowledgment, retrieve reject, suspend, suspend acknowledgment, suspend reject, and user information.

• Call clearing Terminates a call. The following events occur in the call-clearing phase: disconnect, release, release complete, restart, and restart acknowledgment.

• Miscellaneous messages Negotiates network features (supplementary services). Miscellaneous services include congestion control, facility, information, notify, register, status, and status inquiry.

Note

ISDN Layer 3 messages, or Q.931, are carried within ISDN Layer 2 frames, called Q.921. Cisco ISDN equipment allows the administrator to monitor these messages as they occur using various debug commands.

QSIG

The QSIG (Q Signaling) protocol is based on the ISDN Q.931 standard and provides signaling for private integrated services network exchange (PINX) devices. Figure 2-39 shows how different QSIG operations map to the OSI model.

PINX includes everything from PBXs and multiplexers to Centrex. QSIG is implemented on PRI interfaces only. By using QSIG PRI signaling, a Cisco device can route incoming voice calls from a PINX across a WAN to a peer Cisco device, which can then transport the signaling and voice packets to a second PINX. ISDN PRI QSIG voice signaling provides the following benefits:

• Connects the Cisco device with digital PBXs that use the QSIG form of CCS

• Provides transparent support for supplementary PBX services so that proprietary PBX features are not lost when connecting PBXs to Cisco networks

• Provides QSIG support based on widely used ISDN Q.931 standards and the European Telecommunications Standards Institute (ETSI) implementation standards, which include the following specifications:

  - European Computer Manufacturers Association (ECMA)-143 Private Integrated Services Network (PISN) Circuit Mode Bearer Services Inter-Exchange Signalling Procedures and Protocol (QSIG-BC). The ECMA-143 standard addresses the signaling procedures and protocol used for circuit-switched call control at the Q-reference point between private integrated services network exchanges (PINXs) that are interconnected in a PISN.

  - ECMA-142 Private Integrated Services Network (PISN) Circuit Mode 64-kbps Bearer Services Service Description, Functional Capabilities and Information Flows (BCSD). The ECMA-142 standard addresses the service description and control aspects of standardized circuit-mode bearer services.

  - ECMA-165 Private Integrated Services Network (PISN) Generic Functional Protocol for the Support of Supplementary Services Inter-Exchange Signalling Procedures and Protocol (QSIG-GF). The ECMA-165 standard addresses the signaling protocol used for supplementary service control and Additional Network Features (ANFs) at the Q reference point.

DPNSS

British Telecom and selected PBX manufacturers originally developed the Digital Private Network Signaling System (DPNSS) in the early 1980s. It was developed and put into use before the ISDN standards were completed because customers wanted to make use of digital facilities as soon as possible.

DPNSS operates over standard ISDN physical interfaces and is described in four documents:

• BTNR 188: Digital Private Networking Signalling System No 1, Issue 6, January 1995.

• BTNR 188-T: Digital Private Networking Signalling System No 1: Testing Schedule.

• BTNR 189: Interworking between DPNSS1 and other Signalling Systems, Issue 3, March 1988.

• BTNR 189-I: Interworking between DPNSS1 and ISDN Signalling Systems, Issue 1, December 1992.

Note

Cisco Systems supports DPNSS on various gateway platforms, such as the Cisco 2600, 3600, and 5300 series. DPNSS is not a common signaling system but is still in use in various parts of the world.

SIGTRAN

SIGTRAN, as illustrated in Figure 2-40, is a signaling protocol defined in RFC 2719 and RFC 2960. It describes the way the IP protocol carries SS7 messages in a VoIP network. SIGTRAN relies on the Stream Control Transport Protocol at Layer 4 of the TCP/IP protocol stack.

Using SIGTRAN, a service provider may interconnect a private VoIP network to the public switched telephone network (PSTN) and ensure that SS7 signals are conveyed end to end.

Note

SIGTRAN is implemented on Cisco IP Transfer Point (ITP) equipment as well as the Cisco SC 2200 Signaling Controller.

SS7

The ITU-T (formerly the CCITT) developed SS7 in 1981. The primary functions and benefits of SS7 are as follows:

• Fast call setup is handled by high-speed circuit-switched connections.

• PBX transaction capabilities (that is, call forwarding, call waiting, call screening, and call transfer) are extended to the entire network.

• A dedicated control channel exists for all signaling functions.

• Each associated trunk group needs only one set of signaling facilities.

• Information, such as address digits, can be transferred directly between control elements.

• There is no chance of mutual interference between voice and control channel because SS7 is out-of-band signaling.

• Because the control channel is not accessible by the user, possible fraudulent use of the network is avoided.

• Connections involving multiple switching offices can be set up more quickly.

Figure 2-41 depicts an implementation of SS7. As a function of customer networks, SS7 can be implemented as CCS across a telephony network enterprise. Using Cisco equipment, service providers can implement SS7 on their networks. Cisco has developed several solutions that support off-loading IP traffic from public networks and that support the direct connection of network access servers to the PSTN using SS7 links. These solutions utilize the Cisco SC2200, BTS 10200, and AS5x00, giving service providers a proven and cost-efficient SS7 solution for connecting dial-access servers and voice gateways to the PSTN.

The Cisco AS5x00 family provides carrier-class, high-density connectivity for VoIP and dial subscribers. The product set supports a wide range of IP services (including voice) and enables carriers and Internet service providers (ISPs) to cost-effectively support increased subscriber services and an increasing subscriber base.

Signaling System Interoperability

In some implementations, it is necessary to convert from one signaling format to another. Conversion is necessary to allow different systems to signal each other. Figure 2-42 illustrates an example of signal conversion.

The FXS phone is using FXS loop-start signaling to connect to the PBX. The user dials 9 for an outside line, which carries the call on the T1 by using CAS. After the call reaches the CO, it travels via an SS7-signaled circuit to an ISDN switch. The call is then conveyed via Q.931 to the ISDN telephone at the called party location. Other conversion applications exist in voice telephony, and the telephony equipment must have the capability to perform these conversions transparently to the end users.