arp_l3-4_toip-aplicacion_v1.0_20140703
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Telefonía IPTelefonía IPArquitectura y aplicacionesArquitectura y aplicaciones
Fredy Campos [email protected]
Carrera de Ingeniería Electrónica y Telecomunicaciones
Universidad Nacional Tecnológica de Lima Surhttp://www.untecs.edu.pe/portal/
ver 1.0
2014
2014 | Fredy Campos | [email protected] Telefonía IP - Arquitectura y Aplicaciones 2
Objetivos
● Presentar las principales arquitecturas y aplicaciones de telefonía IP
Introducing Voice over IP – Part 1
Fredy [email protected]
NOTE:Resume extracted only for academic use.
Introducing VoIP
CVOICE v6.0—1-4
Unified Communications Architecture
IP telephony
Customer contact center
Video telephony
Rich-media conferencing
Third-party applications
CVOICE v6.0—1-5
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
CVOICE v6.0—1-6
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
CVOICE v6.0—1-7
Components of a VoIP Network
ApplicationServer
Multipoint ControlUnit
CallAgent
IP Phone
IP Phone
VideoconferenceStation
Router orGateway
Router orGateway
Router orGateway
PSTN
PBXIP Backbone
CVOICE v6.0—1-8
Basic Components of a Traditional Telephony Network
BostonSan Jose
EdgeDevices
CO COTie
TrunksTie
Trunks
COTrunks
COTrunks
LocalLoops
LocalLoops
Switch SwitchPBX PBX
PSTN
CVOICE v6.0—1-9
Signaling Protocols
Protocol Description
H.323ITU standard protocol for interactive conferencing; evolved from H.320 ISDN standard; flexible, complex
MGCP IETF 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
CVOICE v6.0—1-10
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.
CVOICE v6.0—1-11
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.
CVOICE v6.0—1-12
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.
CVOICE v6.0—1-13
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.
CVOICE v6.0—1-14
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
CVOICE v6.0—1-15
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
CVOICE v6.0—1-16
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.
Q.931
Q.921SIP
PSTN
CVOICE v6.0—1-17
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.
FXSSCCP
PSTN
SCCP Endpoint
CVOICE v6.0—1-18
VoIP Service Considerations
Latency
Jitter
Bandwidth
Packet loss
Reliability
Security
CVOICE v6.0—1-19
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
CVOICE v6.0—1-20
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
CVOICE v6.0—1-21
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
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
CVOICE v6.0—1-22
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
CVOICE v6.0—1-23
Secure RTP
RFC 3711 Provides:
– Encryption
– Message authentication and integrity
– Replay protection
SRTP Stream
GW1 GW2
S0/0S0/0
CVOICE v6.0—1-24
Summary
The 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.
Introducing Voice Gateways
Introducing Voice over IP – Part 2
Fredy [email protected]
NOTE:Resume extracted only for academic use.
CVOICE v6.0—1-26
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
CVOICE v6.0—1-27
Gateways
Support these gateway protocols:
– H.323
– MGCP
– SIP
– SCCP
Provide advanced gateway functionality
– DTMF relay
– Supplementary services
Work with redundant Cisco Unified Communication Manager
Enable call survivability
Provide QSIG support.
Provide fax or modem services, or both
CVOICE v6.0—1-28
IP WAN
PSTN
San Jose Chicago
MGCPSJ-GW
Denver
Unified Communications Manager Cluster
Unified Communications Manager Express H.323
CHI-GW
SIPDNV-GW
SIP Proxy Server
Deploying Gateways
CVOICE v6.0—1-29
Gateway Hardware Platforms
Modern enterprise models:
Cisco 2800 Series Routers Cisco 3800 Series Routers Cisco Catalyst 6500 Series
CVOICE v6.0—1-30
Gateway Hardware Platforms (Cont.)
Well-known and widely used older enterprise models:
Cisco 1751-V RouterEOS: 03/2007EOL: 03/2012
Cisco 3600 Series PlatformsEOS: 12/2004EOL: 12/2008
Cisco 1760-V RouterEOS: 03/2007EOL: 03/2012
Cisco 2600XM Series RoutersEOS: 03/2007EOL: 03/2012
Cisco 3700 Series RoutersEOS: 03/2007EOL: 03/2012
CVOICE v6.0—1-31
Gateway Hardware Platforms (Cont.)
Special voice gateways:
Cisco VG224 and VG248 Gateways
Cisco AS5300 and AS5400 Series Gateways
Cisco 7200 Series RoutersCisco ATA 186
Cisco 827-4V RouterEOS: 05/2005EOL: 05/2010
Cisco AS5850 Gateway
CVOICE v6.0—1-32
Gateway Hardware Platforms (Cont.)
H.323Cisco Unified
Communications Manager MGCP
SIP SCCP
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 Yes1
Cisco 2600XM Series Router Yes Yes No No3
Cisco 3600 Series Platforms Yes Yes No No3
Cisco 3700 Series Routers Yes Yes No No3
Cisco VG224 Gateway Yes2 Yes2 No Yes
Cisco VG248 Gateway No No No Yes
Cisco AS53XX and AS5400 and AS5850 Cisco Gateways 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 Yes2 Yes2 No Yes2
Cisco 7200 Series Routers Yes No No No1 Conferencing and transcoding only2 FXS only3 DSP farm
CVOICE v6.0—1-33
IP WAN
Supported IP telephony deployment models: Single-site deployment
Multisite WAN with centralized call processing
Multisite WAN with distributed call processing
Clustering over the IP WAN
Headquarters
Branch
Applications
Cisco Unified Communications Manager Cluster
Applications
Cisco Unified Communications
ManagerCluster
PSTN
IP Telephony Deployment Models
CVOICE v6.0—1-34
Single-Site Deployment
SIP or SCCP
Cisco Unified Communications Manager Cluster
PSTN
Cisco Unified Communications Manager servers, applications, and DSP resources at same physical location
IP WAN (if one) used for data traffic only
PSTN used for all external calls
Supports approximately 30,000 IP phones per cluster
WAN
Data
Only
CVOICE v6.0—1-35
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.
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.
CVOICE v6.0—1-36
Multisite WAN with Centralized Call Processing Cisco Unified Communications
Manager Cluster
SIP or SCCP
SIP or SCCP SIP or SCCP
PSTNIP
WAN
Cisco Unified Communications Manager at central site; applications and DSP resources centralized or distributed
IP WAN carries voice traffic and call control signaling
Supports approximately 30,000 IP phones per cluster
Call admission control(limit number of calls per site)
SRST for remote branches
AAR used if WAN bandwidth is exceeded
SRST-capable
SRST-capable
CVOICE v6.0—1-37
Design Guidelines
Minimize delay between Cisco Unified Communications Manager and remote locations to reduce voice cut-through delays.
Use the locations mechanism in Cisco Unified 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.
At the remote sites, use SRST, Cisco Unified 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.
CVOICE v6.0—1-38
Multisite WAN with Distributed Call Processing
Gatekeeper
Cisco Unified Communications Manager and applications located at each site
IP WAN carries intercluster call control signaling
Scales to hundreds ofsites
Transparent use of the PSTN if the IP WAN is unavailable
SIP or SCCP
SIP or SCCP
SIP or SCCP
PSTN IPWAN
GK
Cisco Unified Call Manager
Cluster
Cisco Unified Call Manager
Cluster
CVOICE v6.0—1-39
Design Guidelines
Use a Cisco IOS gatekeeper to provide CAC into and out of each site.
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.
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.
CVOICE v6.0—1-40
Clustering over the IP WAN
Publisher or TFTP server
QoS-Enabled Bandwidth
IP WANIP WAN
<40 ms Round-Trip Delay
SIP or SCCP
SIP or SCCP
Applications and Cisco Unified Communications Managers of the same cluster distributed over the IP WAN
IP WAN carries intracluster server communication and signaling Limited number of sites
CVOICE v6.0—1-41
WAN Considerations
40-ms maximum RTT between any two Cisco Unified 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.
The general rule of thumb for bandwidth is to over-provision and undersubscribe.
QoS-enabled bandwidth must be engineered into the network infrastructure.
CVOICE v6.0—1-42
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.
Supported Cisco IP telephony deployment models are single-site, multisite with centralized call processing, multisite with distributed call processing, and clustering over the IP WAN.
In the single-site deployment model, the Cisco Unified Communications Manager applications and the DSP resources are at the same physical location; the PSTN handles all external calls.
Summary
CVOICE v6.0—1-43
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 call-processing agent, and the IP WAN carries voice traffic but not call control signaling between sites.
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.
Summary (Cont.)
2014 | Fredy Campos | [email protected] Telefonía IP - Arquitectura y Aplicaciones 44
Fredy Campos [email protected]
Carrera Profesional de Ingeniería Electrónica y TelecomunicacionesUniversidad Nacional Tecnológica del Cono Sur de Lima
http://www.untecs.edu.pe/portal/