1 Curso de Telefonía IP para el MTC Sesión 1 Introducción Mg. Antonio Ocampo Zúñiga
2 Conceptos Generales
3 VoIP Essentials Family of technologies Carries voice calls over an IP network VoIP services convert traditional TDM analog voice streams into a digital signal Call from: Computer IP Phone Traditional (POTS) phone
4 Business Case for VoIP Cost savings Flexibility Advanced features: Advanced call routing Unified messaging Integrated information systems Voice security Telephony application services
5 Components of a VoIP Network Application Server Multipoint Control Unit PSTN IP Backbone PBX Call Agent IP Phone Router or Gateway Router or Gateway Router or Gateway IP Phone Videoconference Station
6 Basic Components of a Traditional Telephony Network Edge Devices Tie Trunks CO CO Tie Trunks PBX Switch Switch PBX CO Trunks CO Trunks San Jose Local Loops Local Loops Boston PSTN
7 Signaling Protocols Protocol H.323 MGCP SIP SCCP or Skinny Description ITU standard protocol for interactive conferencing; evolved from H.320 ISDN standard; flexible, complex IETF standard for PSTN gateway control; thin device control IETF protocol for interactive and noninteractive conferencing; simpler, but less mature, than H.323 Cisco proprietary protocol used between Cisco Unified Communications Manager and Cisco VoIP phones
8 H.323 Approved in 1996 by the ITU-T. Peer-to-peer protocol where end devices initiate sessions. Widely used with gateways, gatekeepers, or clients, especially video terminals
9 MGCP (Media Gateway Control Protocol) IETF RFC 2705 developed in Client/server protocol that allows a callcontrol device to take control of a specific port on a gateway. For an MGCP interaction to take place with Communications Manager, you have to make sure that the OS is compatible with Communications Manager version. The PRI backhaul and BRI backhauling concepts are the most powerful concepts to MGCP
10 SIP (Session Initiation Protocol) IETF RFC 2543 (1999), RFC 3261 (2002), and RFC 3665 (2003). Based on the logic of the World Wide Web. Widely used with gateways and proxy servers within service provider networks. Peer-to-peer protocol where end devices (user agents) initiate sessions. ASCII text-based for easy implementation and debugging.
11 SCCP (Skinny Call Control Protocol) Cisco proprietary terminal control protocol. Stimulus protocol: For every event, the end device sends a message to the Cisco Unified Communications Manager. Can be used to control gateway FXS ports. Proprietary nature allows quick additions and changes.
12 Comparing Signaling Protocols H.323 suite: Peer-to-peer protocol Gateway configuration necessary because gateway must maintain dial plan and route pattern. PSTN H.323 Q.921 Q.931
13 Comparing Signaling Protocols (Cont.) MGCP: Works in a client/server architecture Simplified configuration Communications Manager maintains the dial plan PSTN MGCP Q.921 Q.931
14 Comparing Signaling Protocols (Cont.) SIP: Peer-to-peer protocol. Gateway configuration is necessary because the gateway must maintain a dial plan and route pattern. PSTN SIP Q.921 Q.931
15 Comparing Signaling Protocols (Cont.) SCCP Works in a client/server architecture. Simplified configuration. Communications Manager maintains a dial plan and route patterns. PSTN SCCP Endpoint SCCP
16 VoIP Service Considerations Latency Jitter Bandwidth Packet loss Reliability Security
17 Media Transmission Protocols Real-Time Transport Protocol: Delivers the actual audio and video streams over networks Real-Time Transport Control Protocol: Provides out-of-band control information for an RTP flow
18 Media Transmission Protocols crtp: Compresses IP/UDP/RTP headers on low-speed serial links SRTP: Provides encryption, message authentication and integrity, and replay protection to the RTP data
19 Real-Time Transport Protocol GateKeeper GW1 GW2 Provides end-to-end network functions and delivery services for delay-sensitive, real-time data, such as voice and video Runs on top of UDP RTP Stream Works well with queuing to prioritize voice traffic over other traffic
20 Real-Time Transport Protocol GateKeeper GW1 GW2 RTP Stream Services include: Payload-type identification Sequence numbering Time stamping Delivery monitoring
21 Real-Time Transport Control Protocol Define in RFCs 1889, 3550 Provides out-of-band control information for a RTP flow Used for QoS reporting Monitors the quality of the data distribution and provides control information
22 Real-Time Transport Control Protocol Provides feedback on current network conditions Allows hosts involved in an RTP session to exchange information about monitoring and controlling the session Provides a separate flow from RTP for UDP transport use
23 Compressed RTP S0/0 GW1 crtp on Slow-Speed Serial Links RTP Stream S0/0 GW2 RFCs RFC 2508, Compressing IP/UDP/RTP Headers for Low-Speed Serial Links RFC 2509, IP Header Compression over PPP
24 Compressed RTP S0/0 GW1 crtp on Slow-Speed Serial Links RTP Stream S0/0 GW2 Enhanced CRTP RFC 3545, Enhanced Compressed RTP (CRTP) for Links with High Delay, Packet Loss and Reordering Compresses 40-byte header to approximately 2 to 4 bytes
26 Summary VoIP is the family of technologies that allow IP networks to be used for voice applications, such as telephony, voice instant messaging, and teleconferencing. VoIP uses H.323, MGCP, SIP, and SCCP call signaling and call control protocols.
27 Summary Signaling protocol models range from peerto-peer, client server, and stimulus protocol. Configuring voice in a data network requires network services with low delay, minimal jitter, and minimal packet loss. The actual voice conversations are transported across the transmission media using RTP and other RTP related protocols.
28 Voice Gateways
29 Understanding Gateways A gateway connects IP communication networks to analog devices, to the PSTN, or to a PBX Specifically, its role is the following: Convert IP telephony packets into analog or digital signals Connect an IP telephony network to analog or digital trunks or to individual analog stations Two gateway signaling types: Analog Digital
30 Gateways Support these gateway protocols: H.323 MGCP SIP SCCP Provide advanced gateway functionality DTMF relay Supplementary services
31 Gateways Work with redundant Communication Manager Enable call survivability Provide QSIG support. Provide fax or modem services, or both
32 Deploying Gateways Lima Unified Communications Manager Cluster IP WAN Arequipa MGCP LI-GW PSTN Unified Communications Manager Express H.323 AQ-GW Trujillo SIP TR-GW SIP Proxy Server
33 Gateway Hardware Platforms Modern enterprise models: Cisco 2800 Series Routers Cisco 3800 Series Routers Cisco Catalyst 6500 Series
34 Gateway Hardware Platforms (Cont.) Well-known and widely used older enterprise models: Cisco 1751-V Router EOS: 03/2007 EOL: 03/2012 Cisco 1760-V Router EOS: 03/2007 EOL: 03/2012 Cisco 2600XM Series Routers EOS: 03/2007 EOL: 03/2012 Cisco 3600 Series Platforms EOS: 12/2004 EOL: 12/2008 Cisco 3700 Series Routers EOS: 03/2007 EOL: 03/2012
35 Gateway Hardware Platforms (Cont.) Special voice gateways: Cisco VG224 and VG248 Gateways Cisco AS5300 and AS5400 Series Gateways Cisco AS5850 Gateway Cisco 827-4V Router EOS: 05/2005 EOL: 05/2010 Cisco ATA 186 Cisco 7200 Series Routers
36 Gateway Hardware Platforms (Cont.) H.323 Cisco Unified Communications Manager MGCP Cisco 827-4V Router Yes No No No Cisco 2800 Series Routers Yes Yes Yes Yes Cisco 3800 Series Routers Yes Yes Yes Yes Cisco 1751-V and 1760-V Routers Yes Yes No Yes 1 Cisco 2600XM Series Router Yes Yes No No 3 Cisco 3600 Series Platforms Yes Yes No No 3 Cisco 3700 Series Routers Yes Yes No No 3 Cisco VG224 Gateway Yes 2 Yes 2 No Yes Cisco VG248 Gateway No No No Yes Cisco AS53XX and AS5400 and AS5850 Cisco Gateways SIP SCCP Yes No No No Communication Media Module Yes Yes Yes Yes GW Module WS-X6608-x1 and FXS Module WS-X6624 No Yes No Yes Cisco ATA 180 Series Yes 2 Yes 2 No Yes 2 Cisco 7200 Series Routers Yes No No No 1 Conferencing and transcoding only 2 FXS only 3 DSP farm
37 IP Telephony Deployment Models Applications Communications Manager Cluster PSTN Communications Manager Cluster Applications IP WAN Branch Headquarters Single-site deployment Multisite WAN with centralized call processing
38 IP Telephony Deployment Models Applications Communications Manager Cluster PSTN Communications Manager Cluster Applications IP WAN Branch Headquarters Multisite WAN with distributed call processing Clustering over the IP WAN
39 Single-Site Deployment Communications Manager servers, applications, and DSP resources at same physical location IP WAN (if one) used for data traffic only Communications Manager Cluster PSTN SIP or SCCP WAN
40 Single-Site Deployment PSTN used for all external calls Supports approximately 30,000 IP phones per cluster Communications Manager Cluster PSTN SIP or SCCP WAN
41 Design Guidelines Provide a highly available, fault-tolerant infrastructure. Understand the current calling patterns within the enterprise. Use the G.711 codec for all endpoints; DSP resources can be allocated to other functions, such as conferencing and MTP.
42 Design Guidelines Use H.323, SIP, SRST, and MGCP gateways for the PSTN. Implement the recommended network infrastructure for high availability, connectivity options for phones, QoS mechanisms, and security.
43 Multisite WAN with Centralized Call Processing Unified Communications Manager at central site; applications and DSP resources centralized or distributed IP WAN carries voice traffic and call control signaling SRSTcapable SRSTcapable SIP or SCCP PSTN IP WAN Communications Manager Cluster SIP or SCCP SIP or SCCP
44 Multisite WAN with Centralized Call Processing Supports approximately 30,000 IP phones per cluster Call admission control (limit number of calls per site) SRST for remote branches SRSTcapable SRSTcapable SIP or SCCP PSTN IP WAN Communications Manager Cluster SIP or SCCP SIP or SCCP
45 Design Guidelines Minimize delay between Communications Manager and remote locations to reduce voice cut-through delays. Use the locations mechanism in Communications Manager to provide call admission control into and out of remote branches. The number of IP phones and line appearances supported in SRST mode at each remote site depends on the branch router platform.
46 Design Guidelines At the remote sites, use SRST, Communications Manager Express in SRST mode, SIP SRST, and MGCP gateway fallback to ensure call-processing survivability in the event of a WAN failure. Use HSRP to provide backup gateways and gatekeepers.
47 Multisite WAN with Distributed Call Processing Communications Manager and applications located at each site IP WAN carries intercluster call control signaling SIP or SCCP SIP or SCCP PSTN Call Manager Cluster GK IP WAN Call Manager Cluster Gatekeeper SIP or SCCP
48 Multisite WAN with Distributed Call Processing Scales to hundreds of sites Transparent use of the PSTN if the IP WAN is unavailable SIP or SCCP SIP or SCCP PSTN GK IP WAN Call Manager Cluster Gatekeeper SIP or SCCP Call Manager Cluster
49 Design Guidelines Use HSRP gatekeeper pairs, gatekeeper clustering, and alternate gatekeeper support for resiliency. Size the gateway and gatekeeper platforms appropriately per the SRND. Deploy a single WAN codec.
50 Design Guidelines Gatekeeper networks scale to hundreds of sites. Provide adequate redundancy for the SIP proxies. Ensure that the SIP proxies have the capacity for the call rate and number of calls required in the network.
51 Clustering over the IP WAN Publisher or TFTP server <40 ms Round-Trip Delay IP WAN SIP or SCCP QoS-Enabled Bandwidth Applications and Communications Managers of the same cluster distributed over the IP WAN IP WAN carries intracluster server communication and signaling SIP or SCCP
52 Clustering over the IP WAN Publisher or TFTP server <40 ms Round-Trip Delay IP WAN SIP or SCCP QoS-Enabled Bandwidth Limited number of sites SIP or SCCP
53 WAN Considerations 40-ms maximum RTT between any two Communications Manager servers in the cluster Use QoS to minimize jitter for the IP Precedence 3 ICCS traffic. Design network to provide sufficient prioritized bandwidth for all ICCS traffic, especially the priority ICCS traffic.
54 WAN Considerations The general rule of thumb for bandwidth is to over-provision and undersubscribe. QoS-enabled bandwidth must be engineered into the network infrastructure.
55 Summary Gateways connect IP communications networks to traditional telephony networks. There are several types of voice gateways that can be used to meet all kinds of customer needs, from small enterprises to large service provider networks.
56 Summary Telephony deployment models are singlesite, multisite with centralized call processing, multisite with distributed call processing, and clustering over the IP WAN. In the single-site deployment model, the Communications Manager applications and the DSP resources are at the same physical location; the PSTN handles all external calls.
57 Summary The multisite centralized model has a single call-processing agent, applications and DSP resources are centralized or distributed, and the IP WAN carries voice traffic and call control signaling between sites. The multisite distributed model has multiple independent sites, each with a callprocessing agent, and the IP WAN carries voice traffic but not call control signaling between sites.
58 Summary Clustering over an IP WAN provides central administration, a unified dial plan, feature extension to all offices, and support for more remote phones during failover, but places strict delay and bandwidth requirements on the WAN.
CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,
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