sp voice - cisco · voice encapsulation over ip designed to carry real-time traffic on top of ip...

Copyright © 2005, Cisco Systems, Inc. All rights reserved. Printed in USA.Presentation_ID.scr

© 2006 Cisco Systems, Inc. All rights reserved. Cisco ConfidentialPresentation_ID 1

Mehmud Ameen, Consulting Systems Engineer (CSE)Cisco EXPO April 2007

SP Voice


SP Voice Technology

� From Traditional PSTN to SP Packet Telephony

� Cisco Open Packet Telephony (OPT) Model

� VoIP Signaling Protocols

� Overview of Applications Enabled by OPT

� Business Voice Services

� Next Generation IMS Architecture



PSTN Basic Architecture

SS7 Signaling Link

Bearer Trunk

Access Line

SSP = Service Switching Point

SCP = Service Control Point

STP = Signal Transfer Point

PBX = Private Branch eXchange

Trunk Network

SSP

SSP

SSP

SignalingNetwork

Tie Line(Analog)

ISDN PRI (T1/E1)

Analog or ISDN BRI2/4-Wires

T1/E1/T3/E3/SONET/SDH

T1/E1 DS0, V.35

PBX

STPSCP SCP

STP

STP

PBX


Call Setup with SS7 ISUP

IAM

Initial Address Message

ACM

Address Complete Message

ANM

Answer Message

CIC

Circuit Identification Code

CIC X CIC Y

Line SeizedDTMF

IAMPC/CIC X

IAMPC/CIC X

IAMPC/CIC X

IAMPC/CIC Y

IAMPC/CIC Y

Ringing

IAM

IAM

ACMACM

ANMANM

Off-HookDTMF

Ringing

Off Hook

Number AnalysisTrunk (CIC) Selection

Ringback

STPSTP

STP



Traditional PSTN Model Summary

Call Control

Services

Platform Embedded

Call

Logic

Switching

Logic

Service

Logic

Service Layer

IN Voice Services

Call-Control LayerSignaling and Call Control

Service Access Function

Connection Layer

Circuit Selection

Circuit Switching

Accessing Serv

ices SS7 (INAP/TCAP)

Voice Switching

64 kbps

STP

DB DB

STP

X XX X

BillingBilling


SP Packet Telephony Architecture

Cable

ATM/FREthernet

ISDN DSL

QoS-ManagedPacket Core

Flexible Aggregation

Standard Interfaces

New IP-BasedServices

Traditional PSTN Services

Call Signalingand Routing

LL

Rich SignalingCapability

SCP

SS7

Carrier

Interconnects

Efficient BW Utilization

Internet

STP

TDM



Cisco Open Packet Telephony Model

� Responds to requests from the call control plane by instructing it how to treat calls

� Provides connectivity to AIN applications and engraves for development of new IP-based service products

� Responsible for processing call requests

� Directs the connection control plane to establish bearers

� End-to-end call context is maintained

� May initiate a service request to the services plane

� Manages switching logic associated with the bearer channels

� Performs codec transcoding

� Negotiations for connections, bearer switching, SIP/H.323

ServiceLogic

Services PlaneIN Service LogicAAA Number/Address Resolution

CallLogic

Call Control Plane

Signaling and Call ControlService Access FunctionSwitch-Based Service LogicEnd-to-End Voice Services

SwitchingLogic

Connection Control PlaneConnection NegotiationTranscodingBearer SwitchingMedia Control

Open,Standandard

Interfaces

Open,Standandard

Interfaces


OPT Connection Control Plane

Packet Voice Bearer Switching and Transport Technology

SwitchingLogic

Connection Control Plane

Connection NegotiationTranscodingBearer SwitchingMedia Control

Switches, Routers, Gateways, Access Servers



Voice Packetization

DSPG.711

Analog Digital

(Predigitized)

Compressed Packetized

L3 L2

Reverse Process

Packet Transport

IP Network

Sampling (PAM)

Companding (A-Law, Mu-Law)

PCM Coding (PCM G.711)

G.726 ADPCM

G.728 LD-CELP

G.729(AB) CS-ACELP

G.723.1 ACELP/MPMLQ

VoIP (ovL2)

VoATM

(VoFR)


Voice Encapsulation over IP

� Designed to carry real-time traffic on top of IP

� Real-Time Protocol (RTP)—media

� Real-Time Transport Control Protocol (RTCP)—form of signaling between RTP termination points

Watches the quality of underlying infrastructure

� RFC1889 and 1890

RTP

RTCP

Voice Payload RTP UDP IP

Variable 12 8 20

L2

Cisco IOS® RTP UDP Port Range= <16384–32768>

Four Ports Dynamically Allocated per Single

Full-Duplex CallEven-Numbered Ports

Odd-Numbered Ports

Routing/Addressing

Ports Multiplexing/(CRC)

Sequence Numbers

Payload Type Identification

Timestamps

IP Network

Cisco IOS Voice Gateway

Cisco IOS Voice Gateway



Voice QoS Network Requirements

� Low delay

Long delays cause the listener to start to talk before the speaker is finished

� Low delay variation (jitter)

Jitter causes gaps in the speech pattern that cause the quality of voice to sound “jerky”

� Low packet loss

Packet loss causes voice to sound “jerky” and annoying

� Low echo

Listener annoyed by hearing the speaker twice phase-shifted


OPT Call Control Plane

H.323 SIP MGCP

Call Control and Signaling Equipment

Gatekeeper

Media GatewayController

SIP Server

GK

CallLogic

Call Control Plane

Signaling and Call ControlService Access FunctionSwitch-Based Service LogicEnd-to-End Voice Services

Softswitch



One logicalVoIP Gateway

Softswitch Signaling Architecture

Media Gateway

Media Gateway

Signaling Gateway

Signaling Gateway

Media Gateway

Controller

Media Gateway

ControllerMGCMGCGlobal Signaling

Signal Backhaul M

edia GW Control

Media Gateway

Media Gateway

Media Transport

PSTN Signaling

PSTN Bearer s

SIGTRAN MGCPH.248

SS7ISDNQ.Sig

T1/E1/PRIE&M

FXS/FXO

RTP/UDP/IPATM

AAL1,2,5

H.323SIP

ISUP/TCP


MGCP Architecture

Residential Gateway

Access Gateway

TrunkingGateway

Media Gateway Controller

C4/C5

IMTsFXS

PRI

AnalogPhone

PBX

SS7 Link

SS7 Link

MGCP

SS7 Q.931

PacketNetwork

RTP STP



Scope of H.323 Standards

Video CodecH.261, H.263

Call Control

H.225.0

System Control

Receive

Path

Delay

(Sync)

Audio Codec

G.711, G.722,

G.723, G.728,

G.729

RAS ControlH.225.0

UDP

H.245 Control

H.225

Layer

RTP

RTCP

TCP

UDP

IP

Audio I/O

Equipment

User Data Applications

T.120 etc.

System ControlUser Interface

Video I/O

Equipment


H.323 - Signaling Call Flow

H.323 VoIP NetworkH.323 VoIP Network

RTP / RTCP StreamRTP / RTCP StreamRTP / RTCP Stream

Setup

Call Proceeding

Alerting

Connect

H.225

(TCP Port 1720)

H.245 (Dynamic

TCP Ports)

Media (UDP)

Capabilities Exchange

Open Logical Channel

Open Logical Channel Acknowledge

Gatekeeper

Calling Party

PSTN

Called Party

PSTN

SignalingSignaling

MediaMedia



Cisco Multiservice IP-to-IP Gateway

� Reliable Billing

� Network Privacy and Security

� Call Admission and Routing Control

� Protocol Inter-working

• A new VoIP product to serve as a network demarcation pointnetwork demarcation point for:

323 - 323323 - 323

323 - SIP323 - SIP

SIP - SIPSIP - SIP


IP

A

Enterprise #1

Where VoIP Networks Interconnect

SMB

SP 2

GK

Call Control Server Farm

MGX, AS5000XM

PSTN

Residential Users with Linksys CPE

LI CollectionFunction

LI AdministrationFunction

Rich Media Inter-EnterpriseVoIP PSTN Hop-OffResidential with LawfulIntercept

MGX, AS5000XM

Protocol IWF, NAPT, VPN Interconnect, Security, QoS, SLA, Topology Hiding CAC, Transcoding

SBC

SBC

SBC

Protocol Translation, Topology Hiding, Security, Transcoding,

CCM Interworking, RSVP Proxy

SBC

SBC

SBC

Protocol Translation & interworking, Transcoding, Security, Billing

SBCSBC

CSCP



� SIP is an application layer control (signaling) protocol for creating, modifying and terminating multimedia sessions with one or more participants

� SIP entities can communicate via unicast, multicast, via a mesh of unicast relations or a combination of these

� SIP is using UDP, TCP, SCTP or TLS over TCP in IPv4 and IPv6 environments

Session Initiation Protocol


SIP Components

SIP User

Agents

Registrar RedirectLocation

Database

SIP Proxy

SIP

Servers/

Services

REGISTER“Here I am”

INVITE

“I want to talk to another UA

Proxied INVITE

“I’ll handle it for you”

“Where is this

name/phone#?”

3xx Redirection

“They moved, try this address”

SIP User

Agents SIP-GW



SIP Addressing

� Fully-Qualified Domain Names (FQDN)

sip:jdoe.cisco.com

� SMTP-style Domain Names [RFC 2368]

sip:[email protected]

� E.164 style addresses [RFC 2806]

sip:[email protected]; user=phone

user=phone means this is a gateway

(gateway.com is the FQDN of the egress IP gateway)

� Mixed addresses

sip:[email protected]; user=phone

sip:[email protected]


General SIP Packet format

IHL Total Length

Identification Flags Fragment Offset

Header ChecksumProtocolTime to Live

Source Address

Destination Address

PaddingOptions

Version DSCP ECN

IPv4 Header is 20 Bytes

and Binary

Source PortSource Port Destination PortDestination Port

ChecksumChecksumLengthLength

UDP Header is 8 Bytes and Binary(Layer 4 here could also be TCP or SCTP)

SIP Headers in US-ASCII(variable in length per header/per message)SIP Header is Text-based

and variable in length

SIP messages *sometimes* have a message body- a SIP message header indicates the type of body- could be text, data, audio, video or something else

SIP message body is also

variable, but not always

present (depending on

the Message-type)



Jam

es P

olk

20

05

05

03

Call Forking Flow

Invite

Invite

Alice

OfficePhone

OfficePhone

Bob’s

CellphoneCellphoneBob’s

- Biloxi.com Proxy Server is “Transaction Stateful” in order to properly generate Cancels

- If the Proxy is also “Dialog Stateful”, it will receive BYE Request from Alice or Bob

Invite

Invite Invite

Atlanta.com Biloxi.com 180 Rin

ging

180 Ringing

180 Ringing

200 OK

180 Ringing

200 OK

ACK 200 OK

Session Established

CANCEL

CANCEL

RFC 3261

Mobile LaptopMobile LaptopBob’s


Current Cisco SIP Product Portfolio

IP Phones

ATA 186Linksys

PIX Firewall(& IOS FW and NAT)

Load Balancer

Cisco SIP Proxy ServerCisco CSCP

Media Gateways

Softswitch(PGW2200 & BTS 10200)

Unity CPE (CCME, SRST, Unity Express)

Meeting PlaceDynamicsoft

Service Engines

Cisco CallManager



OPT Services Plane

� IN Services

� Unified communications

� IP contact centers

� Calling card

� Debit card

� International callback

� Operator services

� Directory assistance

� Lawful intercept

� Single number

� Billing/radius

� Web-enabled services

TCAP/INAP GKTMP SIP (J)TAPI SMDI

Open Standard Interfaces

ServiceLogic

Services PlaneIN Service LogicAAANumber/Address Resolution

Enables Innovative Services

DBDB


Packet Transit

� Packet network used for simple minute transport

� Investment in packet technology rather than TDM technology for “future proofing” and network utilization

SS7 SS7

PSTNPSTN

IP Network

RTP or PVC/SVC

Local/Transit

Exchange

MG

SLT SLT

MG

PSTN Gateway

MGC MGC

MGCP MGCP

Inter-MGC

Local/Transit

Exchange

SS7SS7

STPSTP

STPSTP

STPSTP

STPSTP

PSTN Gateway



EISUP

PSTN Gateway to H.323

� PSTN gateway faces PSTN—circuit to packet gateway

� SS7 interworking function to H.323

ITSP UsingH.323

RTP

H.323

H.323

H.323

GatekeeperNative H.323 Endpoints

CiscoCallManager

H.323 Gateways

RTP

MGCP

IP Network

SS7

PSTN

Local/Transit

Exchange

MG

PSTN Gateway

MGC

STPSTP

STPSTP Adjunct

SLTSS7


SS7

PSTN

Local/Transit

Exchange

MG

PSTN Gateway

MGC

STPSTP

STPSTP

PSTN Gateway to SIP

� PSTN gateway faces PSTN—circuit to packet gateway

� SS7 interworking function to SIP

SIP

SIP

SIP Proxy

ITSP UsingSIP

SIP Gateways

Native SIPEndpoints

SLTSS7

RTP

RTP

MGCP



PGW 2200PGW 2200

SIP Proxy ServerAS5X00AS5X00

MGCP

SIP

SS7 IMT

AS5X00AS5X00

PSTN

Accounting

ETTxxDSL

Cisco ATA

V V

V V

Cisco End to End Solution

B2 Bredband AB is a market leader in the area of broadband communication in Sweden and offers fiberto the building, LAN-Ethernet and DSL broadband services to households and small companies. It reached the goal to create the worlds first genuine two-way broadband infrastructure with access to data, voice and video.

The company is present in Sweden and Norway.

B2: PSTN Gateway to Residential Broadband


Residential Voice over Broadbandwith Cisco BTS 10200

Reference Transport Infrastructure

IP Network

PSTN

Broadband Network

Residential

GW

RTP/voice

VoIP Infrastructure

AnalogPhones

MGCP/SIP

PSTN

IMT

MGCPCisco

Softswitch

Softphone

IP

phone

ETTx, DSL, Cable

AS5000,MGX8000

Intelligent

NetworkLocal Number

Portability

Lawful

InterceptionBilling Advanced

Services

SS7



Cisco Solutions Architecture Integrated Broadband VoIP Services

AGGREGATIONCPE CORE

12K/CRS-1

Media Server

Network resourcesNetwork resources

7600

TRUNKING

MGX 8000ESR10K

PRIE1/PPP

IAD

AnnouncementServer

LIServer

BRI/PRIAggr. Router

CPE

DSLAM

HFC Plant

uBR7246/uBR10012

MTA / EMTA / SMTA BTS10200

UM Server

SS7 Cable

DSL

RKS Server

LL

ITP

IVR Server

ATA188Linksys

Catalyst

ETTx

V V

V

V

PGW2200

PSTN

CallCall ControlControl

IP

Backbone


VPN VPN

Docsis

SP VoIP Architecture

SoftswitchBTS10200

PSTN

SPA

Management and Self-Administration

ITP

PSTN GatewayPGW2200

VM / UM

Lawful Interception

DHCPTFTPDNS

SS7

CME

MGX8880VXSM

SIP

SCCP

SIP

MGCP

SIPSIGTRAN

AAA

STP

SIP

SAP

SAP

Firewall Internet

V V

ATA188

SIP

SBC

MGCP/NCS

CMTS

PBX

CMTS



Cisco SP BVS SolutionEnabling a Portfolio of Business Voice Services

Customer

Options

Service

ProviderServices

Remote IP Phones with Hosted IP-PBX

Managed Cisco CCME IP-PBX

SP BVS Infrastructure

SP BVS Infrastructure

Site-to-

Site Voice

SiteSite--toto--

Site VoiceSite VoicePSTN

Access

PSTNPSTN

AccessAccessEnhanced IP Services

e.g. XML, Video, IP Conferencing

Enhanced IP ServicesEnhanced IP Servicese.g. XML, Video, IP e.g. XML, Video, IP ConferencingConferencing

Unified

Communications

UnifiedUnified

CommunicationsCommunications

Managed VoIP GW with

TDM-PBX

Managed Cisco IP-PBX

A

Managed IAD w/ Analog Phones

Business

PhoneServices

BusinessBusiness

PhonePhone

ServicesServices

CallAccounting

CallAccounting

PSTNConnectivity

PSTNConnectivity

Regulatory

LNP, LI

Regulatory

LNP, LISecurity,

QoS

Security,

QoSScalabilityScalabilityCall

Routing

Call

Routing

BVSM - Business Voice Services Management

iBVS Platform


Multitenant iBVS Architecture

VV

V V

VV

V V

Business Voice

Services Manager

Business Voice

Services Manager

VOSSManagerSystem Management

VOSSManagerSystem Management

RDBMS &LDAP Directory

BVSM

PGW2200

HSI / GK

VM /UM

DHCPDNS

BAMS /

CNRCiscoWork

s

AS / MGX / SLT / ITP

CCM Cluster

PIX / FWSM

GKGK

GKGK

SS7PRI

ISDNQ.SIGDPNSS

SCCP SCCP

SCCPMGCP

MGCP

MGCP / SCTP

FTP SIPH.323

Customer 1 Customer 2 Customer 3



Next Generation IMS Architecture


IP Multimedia Subsystem (IMS)

� IMS is a framework, a concept and an architecture –NOT a protocol

� SIP is a Signaling protocol (RFC3261 and derivatives) which was adapted by 3GPP and 3GPP2

� 3GPP named SIP interface to the application layer in IMS by the letters ISC (IMS Service Control)

� ISC is one of the three pre-requisites for IMS (together with HSS and CSCF(s)) – as was defined lately in an interesting report by HeavyReading

� Yes – SPs can deploy SIP without IMS



IMS Basic Architecture Components

IP Core

HSS

ISC

Cx

Application Layer

Control Layer

Transport Layer

PSTN / PLMN

Media Gateway

Control Function

(MGCF)

Media Gateway (MGW)

H.248

SIGTRAN

SIP

Breakout Gateway

Control Function (BGCF)

Media Resource Function

Controller (MRFC)

Media Resource Function

Processing (MRFP)

Other VoIP

H.248

Application Server (AS)

Call Session

Control Function (CSCF)

Home Subscriber

Server (HSS)

and Subscriber Location

Function (SLF)


IMS based Services

� Push to Talk

From a Buddy List user interface select one or more users with whom to have 1 way, streamed, individual/group conversations like with CB Radio.

� Push to See

While in Circuit Switched conversational voice use your phone user interface to trigger Live video to the person you are talking to.

� Rich Voice

Enrich the Circuit Switched voice experience by providing a presence enabled Buddy List and network based address book from which conversational voice and other multimedia services can be launched.

� Peer to Peer Gaming

� Collaboration

The ability to share multimedia and applications while in a voice/video call.



High Level Target IMS Architecture


Q and A

sp voice - cisco · voice encapsulation over ip designed to carry real-time traffic on top of ip...

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