Cvoice 6.0 S01 L01

Introducing VoIP

Introducing Voice over IP

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-1

Cisco Unified Communications Architecture  IP telephony  Customer contact center  Video telephony  Rich-media conferencing  Third-party applications

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-2

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

© 2008 Cisco Systems, Inc. All rights reserved.

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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

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-4

Components of a VoIP Network PSTN Application Server IP Backbone

Multipoint Control Unit Call Agent IP Phone

Router or Gateway

PBX

Router or Gateway Router or Gateway

IP Phone Videoconference Station

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-5

Basic Components of a Traditional Telephony Network Edge Devices

Tie Trunks PBX

CO

CO

Switch

Switch

CO Trunks Local Loops

Tie Trunks PBX CO Trunks Local Loops

San Jose

Boston PSTN

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-6

Signaling Protocols Protocol H.323

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

MGCP

IETF standard for PSTN gateway control; thin device control

SIP

SCCP or “Skinny”

© 2008 Cisco Systems, Inc. All rights reserved.

Description

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

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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.

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-8

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. © 2008 Cisco Systems, Inc. All rights reserved.

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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.

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-10

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.

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-11

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.

PSTN H.323

Q.921 Q.931

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-12

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    

PSTN MGCP

Q.921 Q.931

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-13

Comparing Signaling Protocols (Cont.) 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.

PSTN SIP

Q.921 Q.931

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-14

Comparing Signaling Protocols (Cont.) 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. PSTN

SCCP

FXS

SCCP Endpoint © 2008 Cisco Systems, Inc. All rights reserved.

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VoIP Service Considerations  Latency  Jitter  Bandwidth  Packet loss  Reliability  Security

© 2008 Cisco Systems, Inc. All rights reserved.

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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

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-17

Real-Time Transport Protocol 323

H.

GateKeeper

H.3 23

SCC

CP SC

P

GW1

GW2 RTP Stream

 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 © 2008 Cisco Systems, Inc. All rights reserved.

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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

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-19

Compressed RTP cRTP on Slow-Speed Serial Links S0/0 S0/0

GW1

GW2 RTP Stream

 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 © 2008 Cisco Systems, Inc. All rights reserved.

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Secure RTP S0/0 S0/0

GW1

GW2

SRTP Stream

 RFC 3711  Provides: – Encryption – Message authentication and integrity – Replay protection

© 2008 Cisco Systems, Inc. All rights reserved.

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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.

© 2008 Cisco Systems, Inc. All rights reserved.

CVOICE v6.0—1-22

© 2008 Cisco Systems, Inc. All rights reserved.

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