SPECTRA2

VoIP
1 August 4, 2010

Objectives of VoIP
WHAT is VoIP VoIP Network Architecture The VoIP Protocols & Call Flows VoIP and PSTN Interworking

August 4, 2010

What Is VoIP?
Voice over Internet Protocol Also called IP Telephony or Internet Telephony Nowadays, not only Voice but also Video Calls are broken down into IP packets using CODECs, and transmitted over networks as data Packets are re-assembled at the receiving end and the CODEC returns the call to analog form
VoIP Terminal
Microphone /Speaker

Signaling Media (Packet)

VoIP Terminal
CODEC
Analog
Microphone /Speaker

Analog

CODEC

Break & Re-assemble

3 August 4, 2010

Break & Re-assemble

What Do VoIP Terminals Look Like?

VoIP calls can be placed on any device which can run a CODEC:
PCs (using a softphone - Skype, Yahoo Messenger, Vonage, PC-based PBX) A dedicated IP-phone (CISCO, AVAYA) Cell phones: digital cell phones already transmit packetized voice and are moving to more popular VoIP CODECs to support wireless roaming to WiFi networks (VoIP Client on Mobile )
4 August 4, 2010

VoIP Technology
Making a Phone Call Means

Calling someone

Signaling

Call setup or session initiation Also known as Signaling Uses the network control plane (layer 4) Key protocols: SIP, H.323 Requires a Signaling Gateway / Proxy Server

Having a conversation

Media

Real-Time data Streaming: layer 3 packets Uses UDP (packets not resent if lost) Key protocol: RTP Common CODECs: G.711, G.723.1, AMR Requires a Media Gateway

There are VoIP Families: SIP Family and H.323 Family

5 August 4, 2010

How does VoIP Work?

The 1-2-3s of VoIP

1. Compression voice is compressed typically with one of the following codecs, G7.11 64k, G7.29AB 8k, G723.1 6.3k Encapsulation the digitized voice is wrapped in an IP packet Routing the voice packet is routed thru the network to its final destination

2. 3.

These steps happen in reverse at the other end

August 4, 2010

VoIP SIP Network Architecture

SIP-ASP ISUP/ Legacy SS7

SIP Nodes Interworking Nodes

ISUP/ SIGTRAN ISUP/ Legacy SS7

SGW

ISUP/ SIGTRAN

SIP-T/SIP-I/BICC

MGC
SIP
P SI
Registrar

MGC
SIP

SGW
Signaling Media

SI P
Redirect Server

8 CP 24 G H. o/M ac eg M

M eg H. ac 2 4 o/ 8 M GC P

Redirect Server

SIP

Proxy

SIP-SIP Call Domain

RTP RTP

SIP
Proxy

TDM

RT P

TE

SIP

SIP

TE

RTP

MGW

MGW

TDM

Proxy: receive SIP client message and forward it;, Network Access Control, Routing, reliable request retransmission, and security; Redirect Server: Alternate Routing for users, provides next hop(s) information; Registrar: Registration, Session Control, Authentication, Authorization; Application Server (ASP): ???????????????????? MGC (Media Gateway Controller): also called softswitch/call agent; Responsible for Call routing, signaling, call services, billing, address translation; SGW (Signaling Gateway): conversion at transport level between the SS7 based and the IP based, does not interpret the application layer (e.g. MAP, CAP, BICC, ISUP);
7

MGW (Media Gateway): Perform media conversion between VoIP RTP & TDM. August 4, 2010

How To Understand the SIP Architecture

Once upon a time, there was a village of people called the SIPPERs. They made a very special drink called Spectra2. Spectra2 was so good, that the SIPPERs had to put a quota on how much the people could have. So they elected a chief who lived in the Registrar and controlled the quotas of Spectra2 using the AS (Allocation System). To get Spectra2, SIPPERs needed to get permission by Proxy, then had to be guided by Redirect, & granted access by the Chief. The SIPPERs rightly imagined that other villages might like to purchase Spectra2. So they made arrangements with the neighboring villages called ISUPs to trade in Spectra2. They stored the Spectra2 in MGWs (Metal Guarded Warehouses). And extended the road to transport Spectra2 from MGW to MGW. They called the road RTP (Road To Profitability). ISUPs did not speak SIPPER language so the SIPPERs had to build translation stations so the chief could receive and fill the orders. Spectra2 became a global industry. SIPPERs became very rich. However, problems did arise as pirates tried to steal the Spectra2. This required the building of castles, gateways, and controllers to ensure the revenue stream and keep the Spectra2 services running.

ISUP ISUP

ISUP

ISUP SIGTRAN SIP

C P

AS
SIP
SI P

ISUP SIGTRAN

ISUP

SGW MGC Redirect

SIP
G M o/ c ga

SGW MGC Redirect

M G M o/ ac eg

P SI

Registrar Registrar
SIP-SIP Call SIPDomain

TE
RTP

SIP

e M

CP

SIP

SIP

Proxy

RTP

Proxy

TE

RTP

MGW
8 August 4, 2010

MGW

VoIP SIP Protocol Family

Signaling: SIP, Defined by IETF, adopted by 3GPP, 3GPP2, OMA, MSF for IMS
A text based protocol that provides call signaling, registration, status, & control Transport-independent, over UDP, TCP, & SCTP SDP (Session Description Protocol) is embedded in order to share endpoint media Commonly used in xDSL, FTTx, broadband VoIP and enterprise cable VoIP offerings SIP Lots of Extensions based on the SIP basic protocol
UDP TLS TCP IPsec IP DL L1 SIP SCTP

SIP-ASP ISUP/ Legacy SS7

SGW

ISUP/ SIGTRAN

SIP-T/SIP-I/BICC

MGC
SIP
P SI
Registrar

MGC
SIP

ISUP/ SIGTRAN

SGW
Signaling Media

ISUP/ Legacy SS7

SI P
Redirect Server

8 CP 24 G H. o/M ac eg M

M eg H. ac 2 4 o/ 8 M GC P

Redirect Server

SIP

Proxy

SIP-SIP Call Domain

RTP RTP

SIP
Proxy

9 TDM August 4, 2010

RT P

TE

SIP

SIP

TE

RTP

MGW

MGW

TDM

VoIP H.323 Network Architecture

Signaling Media H.323 RTP H.323

H.323 Gatekeeper
P RT

MCU

RTP

H.323 Gatekeeper

23 H.3

RTP H.323 H.323

RTP

H.323

H.323

H.323

MGW
RTP

TE
RTP

TE
RTP

MGW

H.323 Gatekeeper: Virtual switch, translating network address; Admissions control, bandwidth control; Call authorization, bandwidth management; Supplementary services, directory services, call management services MGW (Media Gateway): Perform media conversion between TDM & VoIP RTP; MCU (Multi-point Control Unit): Bridge conferencing connections.
10 August 4, 2010

VoIP H.323 Protocol Family

Signaling: H.323, Defined by ITU-T
A suite of protocols that is based on Q.931 and provides call control, media control, and RAS (Registration, Admission, and Status) Binary protocol H.225.0 H.245 Deployed by many early adopters of VoIP H.225.0 Call Control RAS Signaling Signaling H.225.0: Describe Call Signaling, Media (Audio & Video), streaming; UDP TCP H.245: Control protocol for multimedia for opening & closing channels; IP Data Link H.450: Supplementary Services; L1 H.323 H.235: Security; H.261, H.263, H.264: Video encoding.
Signaling Media

H.323 Gatekeeper
P RT

H.323 RTP

H.323

MCU

RTP

H.323 Gatekeeper

23 H.3

RTP H.323 H.323

RTP

H.323

H.323

H.323

MGW
RTP

TE
RTP

TE
RTP

MGW

11

August 4, 2010

VoIP Common Protocols

H.248/MEGACO

H.248/MEGACO (MEdia GAteway COntrol protocol)

A device control protocol that grew out of MGC Available as a binary or text implementation Instructs MGs to setup and teardown voice calls and manages media resources (available circuits and IP ports) Signals endpoint events to the MG (e.g. off-hook, on-hook)

UDP SCTP IP IP DL DL L1 L1 H.248/MEGACO

MGCP (Media Gateway Control Protocol)

Commonly used in residential cable VoIP networks Text-based protocol that is used to establish sessions on a MG Instructs MGs to setup and teardown voice calls and manages media resources MGCP (available circuits and IP ports) UDP SCTP Signals endpoint events to the MG (e.g. off-hook, on-hook) IP IP
DL DL L1 L1 MGCP
12 August 4, 2010

VoIP Common Protocols

SDP (Session Description Protocol)
Embedded in SIP, H.323, Megaco, and MGCP Provides information on the the capabilities of the endpoints in a session, such as supported codecs

Media: RTP, Defined by IETF

RTP (Real time Transport Protocol):
Designed to carrier media over IP Built to be more reliable than TCP and used in place of TCP

RTCP (Real time Transport Control Protocol):

Adjunct protocol to RTP Provide out of band control information for RTP Provides statistics on an RTP session and sends reports on the session Sender reports provides information on packets sent, delay, jitter, timestamps Receiver reports respond back with jitter, delay, and lost packets

RTP/RTCP UDP IP DL L1 RTP/RTCP

13

August 4, 2010

Protocol Comparisons
H.323
Binary
Based on Q.931

SIP
Text
Parameters can be added easily

Deployed in the first and second generation VoIP solutions ISDN-based signaling facilitates SS7 interworking Designed to support voice and video communications

Gaining global popularity as the VoIP protocol of choice SIP-T/SIP-I provides SS7 interworking Contains a service framework (SIMPLE) for presence, messaging, and video

MGCP
Text Adopted by cable operators for device control
NCS / TGCP

MEGACO/H.248
Text and binary implementations Adopted by 3GPP standards for device control Used in fixed and wireless VoIP Networks

More common in fixed line VoIP networks

14 August 4, 2010

Call Basics
1. Registration All VoIP implementations require a device or user to register with the network for security
Register Function

4. Calls to the PSTN are made through the MGC and media is converted through the MG SSP
MGC

4 4

PSTN

IP Network
This is called RAS in H.323 and the registrar in SIP 1 1 3 3
MG

4 4 Voice is received over traditional T1/E1 circuit

3 3

4 4 4 4

2 2 2. Calls can be made direct to another IP phone. 3. Or calls can be routed to a softswitch, then to the called party
15 August 4, 2010

Voice is sent over RTP

SIP Call Flow

User Agent1 SIP Server: Registrar Proxy Redirect User Agent2

Register 200 OK Invite Invite 180 Ringing 200 OK 200 OK ACK ACK RTP Bye 200 OK
16 August 4, 2010

SIP TLS UDP SCTP TCP IP DL L1 SIP

Bye 200 OK

Megaco Call Flow

MGC MG

Add AddReply Modify ModifyReply

RTP stream is initiated from the endpoint
H.248/MEGACO UDP SCTP IP IP DL DL L1 L1 H.248/MEGACO

RTP Subtract SubtractReply

17

August 4, 2010

MGCP Call Flow

MGC MG

CRCX Create Connection 200 OK MDCX Modify Connection 200 OK

RTP stream is initiated from the endpoint
MGCP UDP SCTP IP IP DL DL L1 L1 MGCP

RTP DLCX Delete Connection 250 OK

18

August 4, 2010

A Simple Voice Call in a VoIP Environment

SS7
MGC MG

SIP

MGCP

Media

SSP /LE

Invite SIP signaling provides call control in the IP domain

CRCX MGCP provides device control for setup and tear down on voice session and resource management

IAM SS7 ISUP provides call signaling and media control with device control embedded in the class 5 switch

SDP (embedded in SIP and MGCP) defines media set capabilities and codec selection for the RTP session RTP provides the transport and delivery of packetized real-time media. RTCP monitors quality and reports information about participants in an open RTP session.

19

August 4, 2010

VoIP and PSTN Interworking

VoIP can interworking with different networks VoIP SIP PSTN VoIP H.323 VoIP PSTN VoIP SIP VoIP SIP VoIP SIP VoIP H.323 VoIP Mobile CS Network VoIP IMS (IP Multimedia Subsystem)

20

August 4, 2010

How to Understand the PSTN & VoIP Interworking?

ISUP ISUP ISUP SIP SIP ISUP ISUP ISUP

AS
ISUP/ SIGTRAN ISUP/ SIGTRAN
SIP
SIP

SGW MGC
M eg H. a c 24 o/ 8 M G CP

SGW MGC
8 CP 24 G H. o/M ac eg M

Redirect
SIP

P SI

Registrar Registrar
SIPSIP-SIP Call Domain

SI P

Redirect
SIP

TE

SIP

SIP

TDM

RTP

Proxy

RTP

Proxy

TE

RTP

TDM

MGW

MGW

Transport Convergence SS7/ PSTN

ISUP Legacy

Signaling Convergence

SGW

ISUP SIGTRAN
4 H.2 8

MGC

SIP-T/SIP-I BICC

VoIP
RTP

TDM

MGW
Media Convergence

21

August 4, 2010

SIP VoIP and PSTN Interworking

There are 3 signaling protocols for SIP VoIP and PSTN Interworking:
Bearer Independence Call Control (BICC) Define by ITU-T, ISUP extension Separate Call Control and Bearer Connection Control
- BICC Capability Set 1 for control ATM - BICC Capability Set 2 for control IP

SIP for Telephony (SIP-T) Defined by IETF Transport ISUP Message across IP network as attachment of SIP message

SIP Interworking (SIP-I) Define by ITU-T Mapping between SIP and ISUP messages

SIP-ASP ISUP/ Legacy SS7

SGW

ISUP/ SIGTRAN

SIP-T/SIP-I/BICC

MGC
SIP
P SI
Registrar

MGC
SIP

ISUP/ SIGTRAN

SGW
Signaling Media

ISUP/ Legacy SS7

SI P
Redirect Server

8 CP 24 G H. o/M ac eg M

M eg H. ac 2 4 o/ 8 M GC P

Redirect Server

SIP

Proxy

SIP-SIP Call Domain

RTP RTP

SIP
Proxy

TDM
22

RT P

TE

SIP

SIP

TE

RTP

MGW
August 4, 2010

MGW

TDM

Network Elements For PSTN & VoIP Interworking

Signaling Gateway:
Provides interworking of signaling between PSTN & IP, ISUP/Legacy <->ISUP/SIGTRAN. Often deployed in groups of two or more to ensure high availability.

Media Gateway:
Converge (Compress/Decompress and Packetize/Depacketize) the voice data between PSTN and the IP network.

Media Gateway Controller:

Handles the registration & management of resources at the Media Gateway(s). Converge ISUP and SIP Signaling Protocols Also called a Softswitch.
SIP-ASP ISUP/ Legacy SS7 ISUP/ SIGTRAN SIP-T/SIP-I ISUP/ SIGTRAN

SGW

MGC
SIP
P SI
Registrar

MGC
SIP

SGW
Signaling Media

ISUP/ Legacy SS7

SI P
Redirect Server

8 CP 24 G H. o/M ac eg M

M eg H. ac 2 4 o/ 8 M GC P

Redirect Server

SIP

Proxy

SIP-SIP Call Domain

RTP RTP

SIP
Proxy

TDM
23

RT P

TE

SIP

SIP

TE

RTP

MGW
August 4, 2010

MGW

TDM

SIGTRAN SIGnaling TRANsport

Defined by IETF SIGTRAN Working Group Reliable Transport SS7 over IP networks. The Architecture identifies two components:
A common Transport Protocol for the SS7 protocol layer being carried

An Adaptation Module to emulate lower layers of the protocol.

Functions:
Transfer Signaling over IP Flow Control, Congestion Control, In-sequence delivery of signaling messages Identification of the originating and terminating signaling points & voice circuits Error Detection, Retransmission, Outages Recovery and other error correction Detection of the status of peer entities (e.g., in service, out-of-service, etc.) Security mechanisms to protect the integrity of the signaling information

ISUP over SIGTRAN

24 August 4, 2010

SGW

ISUP over Legacy SS7

How To Understand SIGTRAN?

From Legacy SS7 to SIGTRAN
Step1: Determine which Layer to be Change to xUA
Any layers except MTP1
Note: One exception: M2PA is not xUA

M3UA: Change at MTP3 Layer ISUP MTP3 MTP2 MTP1

Upper layer No Change Change to M3UA Change to MTP3 Change to SCTP/IP Change to MTP2/MTP1

ISUP M3UA SCTP IP

Step2: Change the selected Layer to the corresponding xUA

MTP2 -> M2UA or M2PA MTP3 -> M3UA SCCP -> SUA TCAP -> TUA

SUA: Change at SCCP Layer APP. TCAP SCCP MTP3 MTP2 MTP1
Change to SCTP/IP Change to MTP3/MTP2/MTP1 Change to SUA Change to SCCP Upper layers No Change

APP. TCAP SUA SCTP IP

Step3:
The upper layer(s) of the selected layer remain no change The lower layers of the selected layer are changed to SCTP/IP;

From SIGTRAN to Legacy SS7 Reverse the procedures

25 August 4, 2010

SS7 Family and SITRAN

SS7 Family

SS7 Over E1/T1

SS7 Over IP

SS7 Over ATM

B IS U P

Applications TCAP SCCP MTP3 MTP2 MTP1 SS7 over E1/T1

M2UA

TCAP

TUA SUA M3UA M2PA ISDN IUA DPNSS /DASS DUA V5 V5UA

M TP -3b SSCF SSCOP AAL 5 ATM L1 B IS U P

ISUP TUP
ISUP ISUA SCCP MTP3 MTP2

SIGTRAN
SS7 Apps SS7 Apps ISUP TUP TCAP SCCP MTP3 SUA MTP3 M2UA SCTP IP DL L1 M2UA SIGTRAN TUA SCTP IP DL L1 TUA IUA SCTP IP DL L1 IUA DUA SCTP IP DL L1 DUA ISUP TUP TCAP TCAP SCCP ISDN/ QSIG DPNSS /DASS SS7 Apps TCAP TUP

SS7 Apps ISUP TCAP SCCP M3UA SCTP IP DL L1 M3UA

ISUP

V5

M2PA SCTP IP DL L1 SUA SCTP IP DL L1 M2PA

ISUA SCTP IP DL L1 ISUA

V5UA SCTP IP DL L1 V5UA

26

August 4, 2010

SCTP Stream Control Transmission Protocol

Glue SIGTRAN Adaptation protocol and IP together Acknowledged error-free non-duplicated transfer of signaling info. In-sequence delivery of messages within multiple streams Optional bundling of multiple messages into a single SCTP packet

SS7 Apps TCAP SCCP ISUP TUP

SS7 Apps TCAP SUA SCTP IP DL L1 SUA

SS7 Apps TCAP ISUP TUP SCCP MTP3 M2PA SCTP IP DL L1 M2PA

SS7 Apps TCAP ISUP TUP TCAP SCCP MTP3 M2UA SCTP IP DL L1 M2UA SIGTRAN TUA SCTP IP DL L1 TUA IUA SCTP IP DL L1 IUA DUA SCTP IP DL L1 DUA ISUA SCTP IP DL L1 ISUA V5UA SCTP IP DL L1 V5UA ISDN/ DPNSS QSIG /DASS ISUP V5

M3UA SCTP IP DL L1 M3UA

27

August 4, 2010

M2UA MTP2 User Adaptation Layer

Transporting SS7 MTP2 user (i.e. MTP3) signaling messages over IP Using SCTP Provides the equivalent set of services to its users as MTP Level 2 provides to MTP Level 3. Used between the Signaling Gateway and Media Gateway Controller

SS7 Apps TCAP SCCP ISUP TUP

SS7 Apps TCAP SUA SCTP IP DL L1 SUA

SS7 Apps TCAP ISUP TUP SCCP MTP3 M2PA SCTP IP DL L1 M2PA

SS7 Apps TCAP ISUP TUP TCAP SCCP MTP3 M2UA SCTP IP DL L1 M2UA SIGTRAN TUA SCTP IP DL L1 TUA IUA SCTP IP DL L1 IUA DUA SCTP IP DL L1 DUA ISUA SCTP IP DL L1 ISUA V5UA SCTP IP DL L1 V5UA ISDN/ DPNSS QSIG /DASS ISUP V5

M3UA SCTP IP DL L1 M3UA

28

August 4, 2010

M2PA MTP2 User Peer-to-Peer Adaptation Layer

A Sigtran protocol for transporting SS7 MTP2 user part signaling messages (i.e. MTP3) over IP using SCTP. Used to support full MTP3 message handling and network management between any two SS7 nodes communicating over an IP network. . M2PA can be used
between a signaling gateway and a media gateway controller, between a signaling gateway and an IP signaling point, between two IP signaling points..
SS7 Apps ISUP TUP TCAP SUA SCTP IP DL L1 SUA SS7 Apps TCAP ISUP TUP SCCP MTP3 M2PA SCTP IP DL L1 M2PA SIGTRAN SS7 Apps TCAP ISUP TUP TCAP SCCP MTP3 M2UA SCTP IP DL L1 M2UA TUA SCTP IP DL L1 TUA IUA SCTP IP DL L1 IUA DUA SCTP IP DL L1 DUA ISUA SCTP IP DL L1 ISUA V5UA SCTP IP DL L1 V5UA ISDN/ DPNSS QSIG /DASS ISUP V5

SS7 Apps TCAP SCCP

M3UA SCTP IP DL L1 M3UA

29

August 4, 2010

M3UA MTP Level 3 User Adaptation Layer

Defined by the IETF sigtran Working Group For transporting MTP3 user part signaling messages (e.g., ISUP, TUP, SCCP) over IP using SCTP. M3UA is used:
between a signaling gateway and a media gateway controller or IP telephony database. The signaling gateway receives SS7 signaling using MTP as transport over a standard SS7 link.
SS7 Apps TCAP SCCP ISUP TUP SS7 Apps TCAP SUA SCTP IP DL L1 SUA SS7 Apps TCAP ISUP TUP SCCP MTP3 M2PA SCTP IP DL L1 M2PA SIGTRAN SS7 Apps TCAP ISUP TUP TCAP SCCP MTP3 M2UA SCTP IP DL L1 M2UA TUA SCTP IP DL L1 TUA IUA SCTP IP DL L1 IUA DUA SCTP IP DL L1 DUA ISUA SCTP IP DL L1 ISUA V5UA SCTP IP DL L1 V5UA ISDN/ DPNSS QSIG /DASS ISUP V5

M3UA SCTP IP DL L1 M3UA

30 August 4, 2010

SUA SCCP User Adaptation Layer

Defined by the IETF SIGTRAN Working Group For transporting SS7 SCCP (Signaling Connection Control Part) user part signaling messages (e.g., TCAP) over IP using the SCTP SUA is used
between a signaling gateway and an IP signaling endpoint between IP signaling endpoints.

SUA supports
SCCP unordered and in-sequence connectionless services; Bi-directional connection-oriented services with/without flow control
SS7 Apps TCAP SCCP ISUP TUP SS7 Apps TCAP SUA SCTP IP DL L1 SUA SS7 Apps TCAP ISUP TUP SCCP MTP3 M2PA SCTP IP DL L1 M2PA SIGTRAN SS7 Apps TCAP ISUP TUP TCAP SCCP MTP3 M2UA SCTP IP DL L1 M2UA TUA SCTP IP DL L1 TUA IUA SCTP IP DL L1 IUA DUA SCTP IP DL L1 DUA ISUA SCTP IP DL L1 ISUA V5UA SCTP IP DL L1 V5UA ISDN/ DPNSS QSIG /DASS ISUP V5

M3UA SCTP IP DL L1
31 August M3UA 4, 2010

Thank You

32

August 4, 2010

SIP Exercise Origination for Tester

33

August 4, 2010

SIP Exercise Termination for Tester

34

August 4, 2010