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Cisco Voice over IP • Chapter 9 1009 Signaling is used for signaling from the PBX to the router, signaling between routers, and sig- naling from the router to the PBX. Without these procedures, calls would not be possible. Signaling Between Routers and PBXs When signaling from PBX to router, lifting the handset produces an off-hook condition.The connection appears as a trunk line to the PBX, which signals the router to seize the trunk.The PBX then forwards the dialed digits to the router in the same manner the digits would be for- warded to a telephone company switch or another PBX.The signaling interface may be any of the common signaling methods used to seize a trunk line, such as FXS, FXO, E&M, or T-1/E-1 signaling. As you can see in Figure 9.25, the PBX seizes a trunk line to the router and forwards the dialed digits. Within the router, the dial plan maps the dialed digits to an IP address and initiates a Q.931 call establishment request to the remote peer router that is indicated by an IP address (Figure 9.26).This control channel is used to set up the Real-time Transport Protocol (RTP) audio streams, and the RSVP protocol may be used to request a guaranteed QoS. When the remote router receives the Q.931 call request, it signals a line seizure to the PBX. After the PBX acknowledges this seizure, the router forwards the dialed digits to the PBX and signals a call acknowledgment to the originating router. Figure 9.27 shows this line seizure. www.syngress.com Figure 9.25 PBX-to-Router Signaling Internet PBX PBX Trunk Signaling 555-1212 Figure 9.26 Router-to-Router Signaling Internet PBX PBX Q.931 H.323 Agent H.323 Agent 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1009 1010 Chapter 9 • Cisco Voice over IP VoIP Protocols With VoIP signaling, end stations perform signaling and session establishment.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 teardown between H.323 agents or end stations. RTCP provides reliable information transfer once the audio stream has been established. A transport protocol such as TCP carries the signaling channels between end stations. RTP, which uses UDP, transports the real-time audio stream. RTP uses UDP since it has lower delay than TCP and voice traffic tolerates low levels of loss and cannot effectively exploit retransmission. H.245 control signaling negotiates channel usage and capabilities. H.245 provides for capabil- ities’ exchange between endpoints so that CODECs and other parameters related to the call are agreed upon by the endpoints. It is within H.245 that the audio channel is negotiated.Table 9.7 depicts the relationship between the ISO reference model and the protocols used in IP voice. We will discuss several of these protocols in detail in the next few sections. Table 9.7 ISO Reference Model and H.323 Standards ISO Protocol Layer 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. H.323 Standard and Protocol Stack The H.320 series of standards was defined for ISDN videophones and videoconferencing sys- tems. H.323 is an ITU-T set of standards that defines the components, protocols, and procedures necessary to provide multimedia (audio, video, and data) communications over IP networks. It is probably the most important standard for packetized voice technology. H.323 provides a method to enable other H.32x-compliant products to communicate. In addition to control and call setup standards, H.323 encompasses protocols for audio, video, and data as follows: www.syngress.com Figure 9.27 Router-to-PBX Signaling Internet PBX PBX Q.931 H.323 Agent H.323 Agent Trunk Signaling 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1010 Cisco Voice over IP • Chapter 9 1011 ■ Audio The compression algorithms H.323 supports for audio are all proven ITU stan- dards (G.711, G.723, and G.729). All H.323 terminals must have at least one audio CODEC support specified by G.711. ■ Video Optional. Any video-enabled H.323 terminal must support the ITU-T H.261 encoding and decoding recommendation. (H.263 is optional.) ■ Data H.323 references the T.120 specifications for data conferencing, which addresses point-to-point and multipoint data conferences. It provides interoperability at the appli- cation, network, and transport levels. Figure 9.28 shows the roles and interoperability of the various H.323 protocols. H.323 is a suite of protocols that provide end-to-end call functionality in a converged network.The H.323 protocol relies heavily on the services provided by other protocols such as TCP, IP, and UDP as well as RTP.The protocols that make up the H.323 protocol are Registration, Admission, and Status (RAS), H.245, and H.225. H.225 H.225 establishes and controls calls between two H.323 endpoints, functions that the ITU Q.931performs for ISDN. Q.931 uses 22 messages, and 29 in case of Q.932. H.225 adopted a subset of Q.931 messages and parameters such as alerting, call processing, connect, setup, release complete, status, status inquiry, and facility (Q.932). H.245 H.245 control signaling negotiates channel usage and capabilities. H.245 exchanges end-to-end control messages managing the operation of the H.323 endpoint. Control messages carry infor- mation related to: www.syngress.com Figure 9.28 H.323 Protocol Interoperability H.245 Control H.225 Call Control H.225 RAS Control Audio Codec G.711 G.722 G.728 G.729 G.732 Real-time Transport Protocol (RTP) Video Codec H.261 H.263 User Data Applications T-120 Video I/O Equipment Audio I/O Equipment System Control/ User Interface RTCP RAS Audio/Video Control TCP/UDP IP LAN 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1011 1012 Chapter 9 • Cisco Voice over IP ■ Capabilities exchange ■ Opening and closing logical channels used to carry media streams ■ Flow control messages ■ General commands After call setup, all communications are over logical channels. H.245 defines procedures for mapping logical channels. Logical channel 0 is for H.245 control for the duration of the call while multiple logical channels of varying types, such as video, data, voice, are allowed for a single call. Real-Time Transport Protocol RTP provides end-to-end network transport functions suitable for applications transmitting real- time data such as audio, video, or simulation data, over multicast or unicast network services. It is used to transport data via UDP. RTP does not address resource reservation and does not guarantee QoS for real-time services. It is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers.The protocol supports the use of RTP-level translators and mixers. RTCP provides a control transport for RTP by providing feedback on the quality of data dis- tribution and carries a transport-level identifier for an RTP source used by receivers to synchro- nize audio and video. Registration, Administration, and Status RAS is a protocol used between endpoints (terminals and gateways) and gatekeepers to perform registration, admission control, bandwidth changes, and status and to disengage endpoints from gatekeepers. RAS uses UDP port 1719. A solid understanding of H.323 components, their functions, and their importance is paramount. All devices that fall within the H.323 protocol stack can be categorized as one of four types of devices.These device types are terminals, gateways, gatekeepers, and multipoint control units (MCUs). H.323 Terminals (Endpoints) H.323 terminals provide the user interface for real-time, two-way multimedia communications. All endpoints must support voice communications and, optionally, video or data communications. For voice, the H.323 terminal is generally an IP telephone. H.323 is deployed as software such as Microsoft NetMeeting . In order to qualify as an H.323 terminal, the device in question must have the following three items: ■ A network interface ■ Audio CODECs ■ H.323 software www.syngress.com 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1012 Cisco Voice over IP • Chapter 9 1013 H.323 terminals must support audio (G.711 is mandatory, and G.723.1 and G.729 are recom- mended for networks of low bandwidth). Video and data support is optional; H.261 is mandatory when video is supported. H.245 and H.225 are required for control functions, and RTP is required for sequencing media packets. H.323 Gateways Gateways translate communications between H.323 and non-H.323 entities (for instance, between H.323 terminals and telephones on the circuit-switched network).They provide call control functions such as address translation and bandwidth management. H.323 gateways enable H.323 networks to communicate with other networks such as the PSTN or PBXs. Gateways provide translation and call control between dissimilar networks. Encoding, protocol translation, and call control mappings occur in gateways between two endpoints. Gateways provide many functions, including: ■ Translating protocols Allows the PSTN and the H.323 network to set up and tear down calls. ■ Converting information formats Enables different networks to freely exchange information such as speech and video. ■ Transferring information Transfers information between dissimilar networks. Gateway functionality is generally provided by a router, such as the 2600 or 3600 series, a Catalyst gateway module such as the 6000 T-1 gateway module, or dedicated gateway devices such as the VG200 and DT-24+. H.323 Gatekeepers Gatekeepers perform call control and policy administration for registered H.323 endpoints. Gatekeepers in H.323 networks are optional. If present, it is mandatory that endpoints use their ser- vices.The H.323 standards define several mandatory services that the gatekeeper must provide: ■ Address translation Translate an alias address into a transport address, which is a PSTN-based phone calling a phone on an IP network (an E.164 number such as 555- 555-2121 will be translated into an IP network address such as 192.168.12.78). ■ Admissions control Defines RAS messages to authorize network access. Does not define the rules or policies used to authorize access to network resources.To do so, the gatekeeper can interface with an existing authorization mechanism. ■ Bandwidth control and management Determines if there is no bandwidth available or no additional bandwidth available for calls requesting increases. Can instruct a call to reduce its bandwidth usage. ■ Zone management An H.323 “zone” is the collection of all components—terminals, gateways, and MCUs—managed by a single gatekeeper. Gatekeeper must provide required functions (for example: address translation, admissions control, bandwidth con- trol) to devices within its zone. www.syngress.com 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1013 1014 Chapter 9 • Cisco Voice over IP The gatekeeper can also perform optional functions such as: ■ Call authorization Authorize or reject a given call; the provider of the H.323 service specifies the reasons for authorization and rejection. ■ Call control signaling Process all call signaling associated with the endpoints regis- tered with it (gatekeeper routed call signaling) or allow the call signaling messages to pass directly between the endpoints. ■ Call management Provide intelligent call management.The call management may be based on address translation functions providing call screening, call forwarding/redirec- tion, and call routing based on time of day, network congestion, or least-cost path. As with gateways, routers are typically incorporated to provide gatekeeper functionality. Multipoint Control Units MCUs provide conference facilities for users who want to conference three or more endpoints together. MCUs do not provide a direct interconnection to the H.323 protocol stack. Rather, they provide a method for H.323 to interconnect voice and videoconferencing. All terminals participating in the conference establish a connection with the MCU. It man- ages conference resources and negotiations between endpoints to determine which audio or video CODEC to use.The MCU might or might not handle the media stream.An MCU has two functional components: ■ Μultipoint controller (MC) Mandatory. Controls where media streams go. Has a reconciliation capability (common mode) and may be located in the terminal, gateway, or gatekeeper. ■ Μultipoint processor (MP) Optional. Mixes, switches, and processes media streams, including some or all of the streams in the conference call (video, data, or audio). H.323 Call Stages The process of establishing and maintaining an H.323 call is a very complex one. We will break this process down into a logical and hierarchical order to show what is occurring at each stage and the requirements and resources used for each stage. H.323 Discovery and Registration The five stages of an H.323 call and details of each of these connections are listed. 1. Discovery and registration 2. Call setup 3. Call-signaling flows 4. Media stream and media control flows 5. Call termination www.syngress.com 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1014 Cisco Voice over IP • Chapter 9 1015 A lot happens within each of these stages; from the time the call is requested to the time it is terminated. Device Discovery and Registration The gatekeeper initiates a “discovery” process to determine the gatekeeper with which the end- point must communicate, as shown in Figure 9.29.This discovery can be either a statically con- figured address or through multicast traffic. Once this is determined, the endpoint or gateway registers with the discovered gatekeeper. Registration is used by the endpoints to identify a zone with which they can be associated (a zone is a collection of H.323 components managed by a single gatekeeper). H.323 can then inform the gatekeeper of the zones’ transport address and alias address. In Figure 9.29: 1. A H.323 gateway (or terminal) sends a request to register (RRQ) message using H.225 RAS on the RAS channel to the gatekeeper. 2. The gatekeeper confirms or denies the registration by sending a registration confirma- tion (RCF) or a “Reject registration” message back to the gateway. Intra-zone Call Placement Once the registration and discovery process is complete, we can place a call. Figure 9.30 shows Gateway X placing a intra-zone call to a terminal connected to Gateway Y, Gateway X sends an admission request (ARQ) message to the gatekeeper requesting permission to place a call to a phone number serviced by Gateway Y. www.syngress.com Figure 9.29 H.323 Gatekeeper Call Control/Signaling: Discovery and Registration H.323 Gateway Gatekeeper 1. RRQ 2. RCF My name or E.164 address. 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1015 1016 Chapter 9 • Cisco Voice over IP In Figure 9.30: 1. Gateway X sends an ARQ message using H.225 RAS to the gatekeeper. 2. Gatekeeper requests direct call signaling by sending an admission confirmation (ACF) to Gateway X. 3. H.323 call setup is initiated. Inter-zone Call Placement The process of placing an inter-zone call is somewhat more complicated and resource–intensive, as the network is larger and divided into multiple zones. In Figure 9.31, Gatekeeper A controls Zone A, and Gatekeeper B controls Zone B. Gateway X (or Terminal X) is registered with Gatekeeper A, and Gateway Y is registered with Gatekeeper B. To place a call to Gateway Y terminal, Gateway X first sends an ARQ message to the gate- keeper requesting permission to make the call. Since Gateway Y is not registered with the gate- keeper in Zone A, we assume that the gateways (terminals) are already registered. Figure 9.31 shows five distinct phases in an inter-zone call placement. 1. ARQ Gateway X requests a connection to Gateway Y from its local gatekeeper. 2. Location request (LRQ) Local gatekeeper for Gateway X does not know the IP address of Gateway Y and is requesting the address from Gateway Y’s local gatekeeper. www.syngress.com Figure 9.30 H.323 Gatekeeper Call Control/Signaling: Call Placement (Intra-zone) H.323 Gateway X Gatekeeper 1. ARQ 2. ACF IP WAN H.323 Gateway Y 3. H.323 Call Setup Figure 9.31 H.323 Gatekeeper Call Control/Signaling: Call Placement (Inter-zone) Zone A H.323 Gateway X Gatekeeper A 1. ARQ 4. ACF IP WAN H.323 Gateway Y 5. Call Setup Zone B Gatekeeper B 2. LRQ 3. LCF H.323 Gateway Z 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1016 Cisco Voice over IP • Chapter 9 1017 3. Location confirm (LCF) Gateway Y’s local gatekeeper responds to Gateway X’s local gatekeeper with the IP address of Gateway Y. 4. ACF The local gatekeeper responds to Gateway X’s request and provides the remote IP address of Gateway Y. 5. Call established The H.323 call is established between Gateway X and Gateway Y. H.323 Call Setup After discovery, registration, and call placement are complete, the H.323 call moves into the call setup stage. At this stage, the gateways are communicating directly to set up the connection. An alternative is gatekeeper-routed call signaling, where all call setup messages traverse the gatekeeper. Figure 9.32 helps us conceptualize this process. The call setup is based on the ITU-Q.931 (H.225 is a subset of Q.931), which provides a means to establish, maintain, and terminate network connections across an ISDN.This process comprises six distinct phases, as shown in Figure 9.32. 1. Gateway X sends an H.225 call-signaling setup message to Gateway Y to request a con- nection. 2. Gateway Y sends an H.225 message back to Gateway X, advising that it may proceed with the call. 3. Gateway Y sends an RAS message (ARQ) on the RAS channel to the gatekeeper to request permission to accept the call. 4. Gatekeeper confirms that the call can be accepted by sending a message (ACF) back to Gateway Y. 5. Gateway Y sends an H.225 message to Gateway X, alerting that the connection has been established. 6. Gateway Y sends an H.225 message to Gateway X, confirming call connection to estab- lish the call. www.syngress.com Figure 9.32 H.323 Call Setup H.323 Gateway X Gatekeeper H.323 Gateway Y 1 2 3 45 6 H.245 Connection Established Q.931 Messages Q.931 Messages RAS Messages 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1017 1018 Chapter 9 • Cisco Voice over IP Logical Channel Setup After call setup, all communications travel over logical channels.The H.245 manages these logical channels. Multiple logical channels of varying types (video, audio, and data) are allowed for a single call. The H.245 Logical Channel Signaling Entity (LCSE) opens a logical channel for each media stream. Channels may be unidirectional or bi-directional. Figure 9.33 helps us visualize how the H.323 utilizes virtual channels. The H.245 control channel is established between Gateway X and Gateway Y. Gateway X uses H.245 to identify its capabilities via a Terminal Capability Set (TCS) message to Gateway Y. The media channel setup flow is as follows: 1. Gateway X exchanges its capabilities with Gateway Y by sending an H.245 TCS mes- sage. 2. Gateway Y acknowledges Gateway X’s capabilities by sending an H.245 TCS Acknowledge message. 3. Gateway Y exchanges its capabilities with Gateway X by sending an H.245 TCS mes- sage. 4. Gateway X acknowledges Gateway Y’s capabilities by sending an H.245 TCS Acknowledge message. 5. Gateway X opens a media channel with Gateway Y by sending an H.245 Open Logical Channel (OLC) message and includes the transport address of the RTCP channel. 6. Gateway Y acknowledges the establishment of the logical channel with Gateway X by sending an H.245 OLC Acknowledge message, including: ■ RTP transport addresses (used to send the RTP media stream) allocated by Gateway Y ■ RTCP address previously received from Gateway X 7. Gateway Y opens a media channel with Gateway X by sending an H.245 OLC message and includes the transport address of the RTCP channel. 8. Gateway X acknowledges the establishment of the logical channel with Gateway Y by sending an H.245 OLC Acknowledge message and includes: www.syngress.com Figure 9.33 Media Channel Setup Data H.245 Audio Video Logical Channels 0 4 2 8 H.323 Gateway X H.323 Gateway Y 253_BDCisco_09.qxd 10/15/03 2:16 PM Page 1018 [...]... support VICs on the Cisco 1750 and 1760 routers.These two chassis require PVDMs to be placed on the motherboard, unlike the Cisco 2600, 3600, and 3700 routers, which have DSP support on the VNMs The Cisco 1760 Modular Access Router The Cisco 1760 has four slots for VICs, and is available in two models .The base model is suited for data networking, but can be upgraded to support voice .The multi-service... wholly Cisco- developed switch, with a router-like IOS .The 3500 Series of switches are fixed configuration switches, and all offer 10/ 100 Ethernet ports and Gigabit Interface Converter (GBIC) ports The 3550-24PWR is the only switch in the 3500 Series that supports inline power www.syngress.com 253_BDCisco_09.qxd 103 2 10/ 15/03 2:16 PM Page 103 2 Chapter 9 • Cisco Voice over IP Catalyst 4x00 Series Switches The. .. Voice over IP • Chapter 9 103 1 NOTE For 3700 platforms, the minimum IOS release is IOS 12.2(8) T for all network modules and VICs 7500 Series Router Configurations The 7500 series high-end routers support voice, video, and data .The Cisco 7500 series includes the Cisco 7505, the Cisco 7507, and the Cisco 7513 with 5, 7, and 13 slots, respectively Cisco 7500 adapters include the two-port T-1 and E-1 high-capacity... to the gatekeeper to request the desired bandwidth 2 The gatekeeper responds with a bandwidth confirmation (BCF) message for the requested bandwidth 3 A logical channel is established between the two gateways with the specified bandwidth 4 A BRQ is sent from the remote router to the gatekeeper to change the bandwidth of the connection 5 The gatekeeper responds to the gateway with a BCF to confirm the. .. receives confirmation that the network can handle the load and provide the requested QoS end to end.To accomplish this task, the network uses a process called admission control www.syngress.com 253_BDCisco_09.qxd 10/ 15/03 2:16 PM Page 103 7 Cisco Voice over IP • Chapter 9 103 7 Cisco IOS uses RSVP and intelligent queuing RSVP is currently in the process of being standardized by the IETF in one of its working... voice-port slot/port (Cisco 175x/1760 and MC3 810) www.syngress.com 253_BDCisco_09.qxd 104 4 10/ 15/03 2:16 PM Page 104 4 Chapter 9 • Cisco Voice over IP 4 Determine the mode in which the jitter buffer will operate for calls on this voice port I Adaptive Adjusts the jitter buffer size and amount of playout delay based on current network conditions.This is the default setting I Fixed Defines the jitter buffer... the packet has arrived at the destination router DSPs can be found as modules inserted onto the motherboard, as on the 1700 series routers, or as slots built onto a VNM that is placed in the router For more information on DSPs, refer back to the section entitled “DSP Provisioning” www.syngress.com 253_BDCisco_09.qxd 10/ 15/03 2:16 PM Page 102 7 Cisco Voice over IP • Chapter 9 102 7 Voice Network Modules... router(config)# voice-port slot/port (Cisco 175x/1760 and MC3 810) 4 Select the appropriate signaling for the interface: router(config-voiceport)# signal [wink-start|immediate|delay-dial] 5 Select the appropriate country codes for call progression signaling .The default is us .The northamerica keyword is for the Cisco MC3 810 multiservice concentrator for versions prior to Cisco IOS Release 12.0(4)T and for... out of the queue, there can be a delay between voice packets that sounds like stuttering speech QoS features can be used to alleviate the effects of jitter by prioritizing the voice traffic over other traffic.You can curb delay using several methods I Queuing Time it takes for a packet to exit the output queue of the device that is routing the data Measured from the time the data is generated into the input... released by the output queue I Network switching Delay across the public network such as a Frame Relay or ATM network I De-jitter Voice traffic works best if there is a constant flow of packets Jitter must be minimized to improve the quality of the conversation De-jitter buffers are utilized on the receiving end to adjust the variable delays into a fixed delay The command that adjusts the Cisco de-jitter . lifting the handset produces an off-hook condition .The connection appears as a trunk line to the PBX, which signals the router to seize the trunk .The PBX then forwards the dialed digits to the router. termination www.syngress.com 253_BDCisco_09.qxd 10/ 15/03 2:16 PM Page 101 4 Cisco Voice over IP • Chapter 9 101 5 A lot happens within each of these stages; from the time the call is requested to the time it is terminated. Device. between the two gateways with the specified bandwidth. 4. A BRQ is sent from the remote router to the gatekeeper to change the bandwidth of the connection. 5. The gatekeeper responds to the gateway