Voip Protocols
Content
Understanding Voice over IP Protocols
Cisco Systems—Service Provider Solutions Engineering February, 2002
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
1
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
2
Why Move to VoIP?
• Cost savings—toll bypass • Open standards—H.323, SIP, MGCP • Multi-vendor interoperability • Integrated IP voice and data networks
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
3
Cisco Packet Voice Architecture
Open Service Application Layer TDM/ Circuit Switch
Switching Network
Line Concentration Call Control Connection Control Features
(JAIN, AIN, TAPI, JTAPI, XML etc.)
Open/Standard Interface
Open Call Control Layer
(SIP, H.323, MGCP, etc.)
Digital Trunk Subsystem
Common Channel Signaling Complex
Administration Maintenance Billing
Open/Standard Interface
Standards-Based StandardsPacket Infrastructure Layer
(IP, ATM)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
4
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
5
Early Adopters— Advanced Services and Toll-Bypass
• Regulatory opportunities allowed for toll-bypass • PC-to-phone, calling-card and international fax services • Cisco-based carriers used standard protocols, but not all carriers implemented standards • Inter-carrier connections had protocol interoperability challenges
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
6
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
7
Making the Rules for VoIP
• IETF (Internet Engineering Task Force)
The community of engineers that standardizes the protocols that define how the Internet and Internet Protocols work. http://www.ietf.org/
• ITU (International Telecommunications Union)
An international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. http://www.itu.int/home/index.html
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
8
Defining the VoIP Protocols
• H.323
An ITU Recommendation that defines “Packet-based multimedia communications systems”. H.323 defines a distributed architecture for creating multimedia applications, including VoIP
• SIP
Defined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia applications, including VoIP
• MGCP
Defined as IETF RFC 2705. MGCP defines a centralized architecture for creating multimedia applications, including VoIP
• H.248
An ITU Recommendation that defines “Gateway Control Protocol”. H.248 is the result of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also known as “Megaco”
• Megaco
Defined as IETF RFC 2885. Megaco defines a centralized architecture
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
9
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
10
H.323 Components
GK
H.323 MCU
Scope of H.323
e
H.323 Gatekeeper
Packet Network
H.323 Terminal
H.323 Gateway
PSTN
ISDN
V.70 Terminal
H.324 Terminal
Speech Terminal
H.320 Terminal
Speech Terminal
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
11
Scope of H.323 Recommendation
Video I/O Equipment
Video Codec H.261, H.263
Audio I/O Equipment User Data Applications T.120, etc.
Audio Codec G.711, G.722, G.723, G.728, G.729
Receive Pain Delay (Sync)
RTP UDP RTCP
H.225 Layer
System Control H.245 Control
IP
TCP
System Control User Interface
Call Control H.225.0 RAS Control H.225.0
UDP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
12
H.323 Signaling
V
V
H.323 Endpoint A
H.323 Endpoint B
Setup Alerting / Connect
H.225 (TCP Port 1720)
Capabilities Exchange / MSD Open Logical Channel Open Logical Channel Acknowledge
H.245 (TCP Dynamic Port)
RTP Stream RTP Stream RTCP Stream
Media (UDP)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
13
Basic H.323 Call
Gatekeeper A
LRQ LCF ACF ACF
Gatekeeper B
RRQ/RCF ARQ
IP Network
H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245
RRQ/RCF
ARQ
V
Gateway A Phone A
RTP
V
Gateway B Phone B
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
14
Deploying H.323 Networks
• • • • Minimizes GK configuration Addition of new zones Addition of new NPAs Addition of new rate centers
DGK
GK
GK
GK
LA GW #1
West Zone
LA GW #2
Chicago GW Midwest Zone
NY GW East Zone
Rate Intra-LATA Rate Center #1 Toll Center #1
Local PSTN
Local PSTN
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
15
MGCP/H.248/Megaco—Architectures
Call Agent SS7
Call Agent
P S T N
IMT
PSTN
PRI Access Gateway
P S T N
MGCP / H.248 / Megaco RTP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
16
Deploying MGCP/H.248/Megaco Networks
OSS CA SS7 Backhaul SS7 SLT MG
MG MG
MGCP and ISDN Backhaul
Billing and Measurement Server
STP PSTN
IMTs
TDM Voice
VoIP ISDN/PRI
Service Provider's TDM Network Service Provider's Packet Network
NAS/VoIP
Traditional TDM Traffic Modem Dial-up Traffic
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
17
SIP Architecture
I N T E L L I G E N T S E R V I C E S 4426_02_2002_c1
Application Services
eMail
LDAP
Oracle
XML
CPL CPL
3pcc
SIP Proxy, Registrar & Redirect Servers SIP SIP SIP SIP User Agents (UA) PSTN
CAS or PRI
RTP (Media) Legacy PBX
© 2002, Cisco Systems, Inc. All rights reserved.
18
SIP Signaling
PSTN SIP VoIP Network
Calling Party INVITE 100 Trying INVITE 100 Trying 180 Ringing 200 OK ACK 180 Ringing 200 OK ACK
PSTN
Called Party
SIP Signaling and SDP Signaling (UDP or TCP)
Signaling
Media (UDP)
Bearer Or Media
RTCP Stream
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
19
SIP Servers/Services
Location Database SIP Servers/ Services “Where is this name/phone#?” 3xx Redirection “They moved, try this address”
Registrar
Redirect
REGISTER “Here I am”
SIP Proxy Proxied INVITE “I’ll handle it for you”
INVITE “I want to talk to another UA
SIP User Agents
SIP User Agents
SIP-GW
20
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
Deploying SIP Networks
PSTN 312 PSTN 212
Chicago POP Central Zone
NY POP East Zone
IP Network
West Zone SF POP
PSTN 415
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
21
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
22
Voice Myths
Myths
• Networks can only be built one way • Networks will only use one protocol • All networks will converge
Facts
• VoIP allows centralized or distributed architectures • H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP networks • Networks will converge to IP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
23
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
24
Interconnecting VoIP Networks
3 2 H.3
SIP
?
MGCP H.248 Megaco
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
25
Connecting VoIP to SS7/C7 Networks
IAM
H.323
H.225 Setup (ANI,DN) Proceeding H.245
MGCP
CRCX ACK SDP
SIP
INVITE ACK SDP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
26
VoIP Interworking Issues
• Service interworking
E.g.: H.450 <-> SIP <-> MGCP
• Media interworking
End-to-end codec negotiation
• Bearer interworking
End-to-end fax, modem, DTMF
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
27
VoIP Interworking
• Bearer level
Modem (relay/passthru) Fax (relay/passthru) T.38 T.37 DTMF (relay/passthru)
• Service translation issues
Call deflection Park/hold
• Signal issues
SDP H.245
• Media level
Codec (negotiation, selection)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
28
Fax and Modem Passthru Mechanisms
• Modem and fax are control mechanisms based on PLL (Phase Locked Loops) • They are both time sensitive • Highly sensitive to packet network impairments:
Jitter Packet loss Delay
• Susceptible to clock slew (clock sync differences between gateways)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
29
Passthru Simplified
Voice Gateway PCM G.711µ DSP G.711µ
G.729
IP Cloud
Voice Gateway G.711µ DSP PCM G.711µ
G.729
4426_02_2002_c1 30
© 2002, Cisco Systems, Inc. All rights reserved.
What Is Modem Passthru?
• It is the transport of modem signals (modulation, error correction and compression) through a packet network using PCM encoded packets
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
31
Modem Passthru (Cont.)
• Modem tone detection (<= V.90) • Switchover signaling • No VAD • EC off • RTP payload redundancy (10ms packetization) RFC2198 (optional)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
32
Modem Passthru Issues
• Consecutive packet drops (loss) cause retrain • Consecutive drops during retrain causes disconnect • Variation of delay (jitter) has quite an effect • Jitter (at 10%) is a conservative estimate— Since jitter mostly impacts performance with packet loss
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
33
What Is Modem Relay?
• Modem relay involves demodulating the modem signal at ingress gateway • Passing this data as packet data to terminating gateway • Re-modulating the data and passes it to the receiving modem
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
34
Fax Relay—T.38
T.30 UDP T.30
PSTN
PSTN
IP
• Real-time • Also called demod/remod • Can be used in H.323/MGCP/SIP signaling • Delivers fax data over UDP streams (uses same RTP port)— reuses voice UDP ports • Fallback to proprietary mode • Method of encoding the T.30 and T.4 into packets
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
35
DTMF
• What is DTMF • Why is it required? and where is it used? • How do you transport it in IP? • DTMF implementation
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
36
DTMF (Cont.)
• In TDM world, all voice traffic is sent as uncompressed 64Kbs PCM streams; anything sent on that circuit is an untouched stream of bits; (e.g., voice speech, modem tones, fax tones, and DTMF digits) • DSP codecs designed to interpret human speech, can distort DTMF tones (machine-tones) • High b/w codecs less likely to distort • Distortion causes problems with voicemail and IVR systems
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
37
DTMF Schemes with VoIP Protocols
H.323
MGCP, H.248, Megaco In-Band
SIP
In-Band
In-Band
Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833
In-Band
Out-ofBand
Cisco RTP, NSE, NTE,RFC2833
RFC2833
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
38
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
39
Summary
• Understand the possibilities and the issues • Avoid protocol/product based bias • Decide on application • Consider market and business drivers • Deploy what’s possible today • Choose signaling protocol depending on services intended to be offered • Many possibilities—stay tuned
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
40
Crystal Ball on VoIP
• All three protocols (or its variations) are here for the long run • Changes/enhancements will be made • IP will be the core
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
41
Reference URLs
• ITU: • IETF: • SIP:
www.itu.org www.ietf.org www.cs.columbia.edu/~hgs/sip/
• H.323: www.packetizer.com/iptel/h323/ • MGCP:www.softswitch.org/asp/techlibrary _protocol.asp?page=techlibrary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
42
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
43
Cisco Systems—Service Provider Solutions Engineering February, 2002
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
1
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
2
Why Move to VoIP?
• Cost savings—toll bypass • Open standards—H.323, SIP, MGCP • Multi-vendor interoperability • Integrated IP voice and data networks
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
3
Cisco Packet Voice Architecture
Open Service Application Layer TDM/ Circuit Switch
Switching Network
Line Concentration Call Control Connection Control Features
(JAIN, AIN, TAPI, JTAPI, XML etc.)
Open/Standard Interface
Open Call Control Layer
(SIP, H.323, MGCP, etc.)
Digital Trunk Subsystem
Common Channel Signaling Complex
Administration Maintenance Billing
Open/Standard Interface
Standards-Based StandardsPacket Infrastructure Layer
(IP, ATM)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
4
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
5
Early Adopters— Advanced Services and Toll-Bypass
• Regulatory opportunities allowed for toll-bypass • PC-to-phone, calling-card and international fax services • Cisco-based carriers used standard protocols, but not all carriers implemented standards • Inter-carrier connections had protocol interoperability challenges
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
6
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
7
Making the Rules for VoIP
• IETF (Internet Engineering Task Force)
The community of engineers that standardizes the protocols that define how the Internet and Internet Protocols work. http://www.ietf.org/
• ITU (International Telecommunications Union)
An international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. http://www.itu.int/home/index.html
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
8
Defining the VoIP Protocols
• H.323
An ITU Recommendation that defines “Packet-based multimedia communications systems”. H.323 defines a distributed architecture for creating multimedia applications, including VoIP
• SIP
Defined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia applications, including VoIP
• MGCP
Defined as IETF RFC 2705. MGCP defines a centralized architecture for creating multimedia applications, including VoIP
• H.248
An ITU Recommendation that defines “Gateway Control Protocol”. H.248 is the result of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also known as “Megaco”
• Megaco
Defined as IETF RFC 2885. Megaco defines a centralized architecture
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
9
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
10
H.323 Components
GK
H.323 MCU
Scope of H.323
e
H.323 Gatekeeper
Packet Network
H.323 Terminal
H.323 Gateway
PSTN
ISDN
V.70 Terminal
H.324 Terminal
Speech Terminal
H.320 Terminal
Speech Terminal
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
11
Scope of H.323 Recommendation
Video I/O Equipment
Video Codec H.261, H.263
Audio I/O Equipment User Data Applications T.120, etc.
Audio Codec G.711, G.722, G.723, G.728, G.729
Receive Pain Delay (Sync)
RTP UDP RTCP
H.225 Layer
System Control H.245 Control
IP
TCP
System Control User Interface
Call Control H.225.0 RAS Control H.225.0
UDP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
12
H.323 Signaling
V
V
H.323 Endpoint A
H.323 Endpoint B
Setup Alerting / Connect
H.225 (TCP Port 1720)
Capabilities Exchange / MSD Open Logical Channel Open Logical Channel Acknowledge
H.245 (TCP Dynamic Port)
RTP Stream RTP Stream RTCP Stream
Media (UDP)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
13
Basic H.323 Call
Gatekeeper A
LRQ LCF ACF ACF
Gatekeeper B
RRQ/RCF ARQ
IP Network
H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245
RRQ/RCF
ARQ
V
Gateway A Phone A
RTP
V
Gateway B Phone B
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
14
Deploying H.323 Networks
• • • • Minimizes GK configuration Addition of new zones Addition of new NPAs Addition of new rate centers
DGK
GK
GK
GK
LA GW #1
West Zone
LA GW #2
Chicago GW Midwest Zone
NY GW East Zone
Rate Intra-LATA Rate Center #1 Toll Center #1
Local PSTN
Local PSTN
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
15
MGCP/H.248/Megaco—Architectures
Call Agent SS7
Call Agent
P S T N
IMT
PSTN
PRI Access Gateway
P S T N
MGCP / H.248 / Megaco RTP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
16
Deploying MGCP/H.248/Megaco Networks
OSS CA SS7 Backhaul SS7 SLT MG
MG MG
MGCP and ISDN Backhaul
Billing and Measurement Server
STP PSTN
IMTs
TDM Voice
VoIP ISDN/PRI
Service Provider's TDM Network Service Provider's Packet Network
NAS/VoIP
Traditional TDM Traffic Modem Dial-up Traffic
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
17
SIP Architecture
I N T E L L I G E N T S E R V I C E S 4426_02_2002_c1
Application Services
LDAP
Oracle
XML
CPL CPL
3pcc
SIP Proxy, Registrar & Redirect Servers SIP SIP SIP SIP User Agents (UA) PSTN
CAS or PRI
RTP (Media) Legacy PBX
© 2002, Cisco Systems, Inc. All rights reserved.
18
SIP Signaling
PSTN SIP VoIP Network
Calling Party INVITE 100 Trying INVITE 100 Trying 180 Ringing 200 OK ACK 180 Ringing 200 OK ACK
PSTN
Called Party
SIP Signaling and SDP Signaling (UDP or TCP)
Signaling
Media (UDP)
Bearer Or Media
RTCP Stream
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
19
SIP Servers/Services
Location Database SIP Servers/ Services “Where is this name/phone#?” 3xx Redirection “They moved, try this address”
Registrar
Redirect
REGISTER “Here I am”
SIP Proxy Proxied INVITE “I’ll handle it for you”
INVITE “I want to talk to another UA
SIP User Agents
SIP User Agents
SIP-GW
20
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
Deploying SIP Networks
PSTN 312 PSTN 212
Chicago POP Central Zone
NY POP East Zone
IP Network
West Zone SF POP
PSTN 415
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
21
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
22
Voice Myths
Myths
• Networks can only be built one way • Networks will only use one protocol • All networks will converge
Facts
• VoIP allows centralized or distributed architectures • H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP networks • Networks will converge to IP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
23
Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
24
Interconnecting VoIP Networks
3 2 H.3
SIP
?
MGCP H.248 Megaco
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
25
Connecting VoIP to SS7/C7 Networks
IAM
H.323
H.225 Setup (ANI,DN) Proceeding H.245
MGCP
CRCX ACK SDP
SIP
INVITE ACK SDP
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
26
VoIP Interworking Issues
• Service interworking
E.g.: H.450 <-> SIP <-> MGCP
• Media interworking
End-to-end codec negotiation
• Bearer interworking
End-to-end fax, modem, DTMF
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
27
VoIP Interworking
• Bearer level
Modem (relay/passthru) Fax (relay/passthru) T.38 T.37 DTMF (relay/passthru)
• Service translation issues
Call deflection Park/hold
• Signal issues
SDP H.245
• Media level
Codec (negotiation, selection)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
28
Fax and Modem Passthru Mechanisms
• Modem and fax are control mechanisms based on PLL (Phase Locked Loops) • They are both time sensitive • Highly sensitive to packet network impairments:
Jitter Packet loss Delay
• Susceptible to clock slew (clock sync differences between gateways)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
29
Passthru Simplified
Voice Gateway PCM G.711µ DSP G.711µ
G.729
IP Cloud
Voice Gateway G.711µ DSP PCM G.711µ
G.729
4426_02_2002_c1 30
© 2002, Cisco Systems, Inc. All rights reserved.
What Is Modem Passthru?
• It is the transport of modem signals (modulation, error correction and compression) through a packet network using PCM encoded packets
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
31
Modem Passthru (Cont.)
• Modem tone detection (<= V.90) • Switchover signaling • No VAD • EC off • RTP payload redundancy (10ms packetization) RFC2198 (optional)
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
32
Modem Passthru Issues
• Consecutive packet drops (loss) cause retrain • Consecutive drops during retrain causes disconnect • Variation of delay (jitter) has quite an effect • Jitter (at 10%) is a conservative estimate— Since jitter mostly impacts performance with packet loss
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
33
What Is Modem Relay?
• Modem relay involves demodulating the modem signal at ingress gateway • Passing this data as packet data to terminating gateway • Re-modulating the data and passes it to the receiving modem
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
34
Fax Relay—T.38
T.30 UDP T.30
PSTN
PSTN
IP
• Real-time • Also called demod/remod • Can be used in H.323/MGCP/SIP signaling • Delivers fax data over UDP streams (uses same RTP port)— reuses voice UDP ports • Fallback to proprietary mode • Method of encoding the T.30 and T.4 into packets
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
35
DTMF
• What is DTMF • Why is it required? and where is it used? • How do you transport it in IP? • DTMF implementation
4426_02_2002_c1
© 2002, Cisco Systems, Inc. All rights reserved.
36
DTMF (Cont.)
• In TDM world, all voice traffic is sent as uncompressed 64Kbs PCM streams; anything sent on that circuit is an untouched stream of bits; (e.g., voice speech, modem tones, fax tones, and DTMF digits) • DSP codecs designed to interpret human speech, can distort DTMF tones (machine-tones) • High b/w codecs less likely to distort • Distortion causes problems with voicemail and IVR systems
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DTMF Schemes with VoIP Protocols
H.323
MGCP, H.248, Megaco In-Band
SIP
In-Band
In-Band
Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833
In-Band
Out-ofBand
Cisco RTP, NSE, NTE,RFC2833
RFC2833
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Topics to Discuss
• History of VoIP • VoIP—Early Adopters • VoIP—Standards and Standards Bodies • VoIP—Making Sense of the Protocols • “The Great Voice Myth” • VoIP—Protocol Challenges • Summary
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Summary
• Understand the possibilities and the issues • Avoid protocol/product based bias • Decide on application • Consider market and business drivers • Deploy what’s possible today • Choose signaling protocol depending on services intended to be offered • Many possibilities—stay tuned
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Crystal Ball on VoIP
• All three protocols (or its variations) are here for the long run • Changes/enhancements will be made • IP will be the core
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Reference URLs
• ITU: • IETF: • SIP:
www.itu.org www.ietf.org www.cs.columbia.edu/~hgs/sip/
• H.323: www.packetizer.com/iptel/h323/ • MGCP:www.softswitch.org/asp/techlibrary _protocol.asp?page=techlibrary
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