cvoice 6.0 s01 l01

© 2008 Cisco Systems, Inc. All rights reserved. CVOICE v6.0—1-1

Introducing Voice over IP

Introducing VoIP


Cisco Unified Communications Architecture

IP telephony

Customer contact center

Video telephony

Rich-media conferencing

Third-party applications


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


Business Case for VoIP

Cost savings

Flexibility

Advanced features:

– Advanced call routing

– Unified messaging

– Integrated information systems

– Long-distance toll bypass

– Voice security

– Customer relationship

– Telephony application services


Components of a VoIP Network

ApplicationServer

Multipoint ControlUnit

CallAgent

IP Phone

IP Phone

VideoconferenceStation

Router orGateway

Router orGateway

Router orGateway

PSTN

PBXIP Backbone


Basic Components of a Traditional Telephony Network

BostonSan Jose

EdgeDevices

CO COTie

TrunksTie

Trunks

COTrunks

COTrunks

LocalLoops

LocalLoops

Switch SwitchPBX PBX

PSTN


Signaling Protocols

Protocol Description

H.323ITU standard protocol for interactive conferencing; evolved from H.320 ISDN standard; flexible, complex

MGCPIETF standard for PSTN gateway control; thin device control

SIPIETF protocol for interactive and noninteractive conferencing; simpler, but less mature, than H.323

SCCP or “Skinny”Cisco proprietary protocol used between Cisco Unified Communications Manager and Cisco VoIP phones


H.323

H.323 suite: Approved in 1996 by the ITU-T.

Peer-to-peer protocol where end devices initiate sessions.

Widely used with gateways, gatekeepers, or third-party H.323 clients, especially video terminals in Cisco Unified Communications.

H.323 gateways are never registered with Cisco Unified Communications Manager; only the IP address is available to confirm that communication is possible.


MGCP

Media Gateway Control Protocol (MGCP): IETF RFC 2705 developed in 1999. Client/server protocol that allows a call-control device to take

control of a specific port on a gateway. For an MGCP interaction to take place with Cisco Unified

Communications Manager, you have to make sure that the Cisco IOS software or Cisco Catalyst operating system is compatible with Cisco Unified Communications Manager version.

MGCP version 0.1 is supported on Cisco Unified Communications Manager.

The PRI backhaul concept is one of the most powerful concepts to the MGCP implementation with Cisco Unified Communications Manager.

BRI backhauling is implemented in recent Cisco IOS versions.


SIP

Session Initiation Protocol (SIP): 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.

SIP gateways are never registered with Cisco Unified Communications Manager; only the IP address is available to confirm that communication is possible.


SCCP

Skinny Call Control Protocol (SCCP): 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.


Comparing Signaling Protocols

H.323 suite: Peer-to-peer protocol

Gateway configuration necessary because gateway must maintain dial plan and route pattern.

Examples: Cisco VG224 Analog Phone Gateway (FXS only) and, Cisco 2800 Series and, Cisco 3800 Series routers.

Q.931

Q.921H.323

PSTN


Comparing Signaling Protocols (Cont.)

MGCP: Works in a client/server architecture Simplified configuration Cisco Unified Communications Manager maintains the dial plan Examples: Cisco VG224 Analog Phone Gateway (FXS only) and,

Cisco 2800 Series and , Cisco 3800 Series routers Cisco Catalyst operating system MGCP example: Cisco Catalyst

6000 WS-X6608-T1 and Catalyst 6000 ws-X6608-E1

Q.931

Q.921MGCP

PSTN



SIP: Peer-to-peer protocol.

Gateway configuration is necessary because the gateway must maintain a dial plan and route pattern.

Examples: Cisco 2800 Series and Cisco 3800 Series routers.

Q.931

Q.921SIP

PSTN



SCCP Works in a client/server architecture.

Simplified configuration.

Cisco Unified Communications Manager maintains a dial plan and route patterns.

Examples: Cisco VG224 (FXS only) and, Cisco VG248 Analog Voice Gateways, Cisco ATA 186, and Cisco 2800 Series with routers FXS ports.

FXSSCCP

PSTN

SCCP Endpoint


VoIP Service Considerations

Latency

Jitter

Bandwidth

Packet loss

Reliability

Security


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

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


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 Works well with queuing to prioritize voice traffic over other traffic Services include:

– Payload-type identification

– Sequence numbering

– Time stamping

– Delivery monitoring

RTP Stream

GW1

GateKeeper

GW2

H.323

SCCPSCCP

H.323

Real-Time Transport Protocol


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

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


RFCs

– RFC 2508, Compressing IP/UDP/RTP Headers for Low-Speed Serial Links

– RFC 2509, IP Header Compression over PPP

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

RTP Stream

GW1 GW2

S0/0S0/0

cRTP on Slow-Speed Serial Links

Compressed RTP


Secure RTP

RFC 3711 Provides:

– Encryption

– Message authentication and integrity

– Replay protection

SRTP Stream

GW1 GW2

S0/0S0/0


Summary

The Cisco Unified Communications System Architecture fully integrates communications by enabling data, voice, and video to be transmitted over a single network infrastructure using standards-based IP.

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.

Signaling protocol models range from peer-to-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.