Académique Documents
Professionnel Documents
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Protocols
Cisco Systems—Service Provider Solutions Engineering
February, 2002
• 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
Open Service
Application Layer
TDM/ (JAIN, AIN, TAPI,
Circuit Switch JTAPI, XML etc.)
Open/Standard
Line Call Control Interface
Switching Network
Digital Trunk
Subsystem
Common Channel
Signaling Complex
Open Call Control Layer
(SIP, H.323, MGCP, etc.)
Administration
Maintenance
Billing
Open/Standard
Interface
Standards-Based
Standards-
Packet Infrastructure Layer
(IP, ATM)
• 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
• 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
• 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
• 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
• 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
H.323
Gateway
PSTN ISDN
V V
H.323
H.323 Endpoint B
Endpoint A Setup
Alerting / Connect H.225 (TCP Port 1720)
RTP Stream
RTP Stream Media (UDP)
RTCP Stream
Gatekeeper A Gatekeeper B
LRQ
LCF
ACF ACF
Phone A Phone B
GK GK GK
West Chicago NY
LA Zone LA GW GW
GW #1 GW #2
Rate Intra-LATA Rate Local Midwest Local East
Center #1 Toll Center #1 PSTN Zone PSTN Zone
P P
IMT
S S
T T
PSTN
N N
PRI
Access
Gateway
PSTN PSTN
SIP VoIP Network
INVITE
Calling Party Called Party
100 Trying INVITE
SIP Signaling
100 Trying
and SDP
Signaling 180 Ringing 180 Ringing Signaling
(UDP or TCP) 200 OK 200 OK
ACK ACK
Bearer Or
Media (UDP) Media
RTCP Stream
SIP
Location Servers/
Registrar Redirect Database Services
“Where is this
name/phone#?”
3xx Redirection
“They moved,
REGISTER SIP Proxy
try this address”
“Here I am”
Proxied INVITE
“I’ll handle it for
INVITE
you”
“I want to talk
to another UA
SIP User
Agents SIP User
Agents SIP-GW
NY
Chicago
POP
POP
Central Zone East Zone
IP Network
West Zone
SF
POP
PSTN
415
• 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
Myths Facts
• 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
23
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.225 Setup (ANI,DN)
Proceeding
H.323
H.245
CRCX
ACK
MGCP
SDP
INVITE
SIP ACK
SDP
• Service interworking
E.g.: H.450 <-> SIP <-> MGCP
• Media interworking
End-to-end codec negotiation
• Bearer interworking
End-to-end fax, modem, DTMF
T.38 Park/hold
Codec (negotiation,
selection)
PCM G.711µ
G.711µ DSP
G.729
IP
Cloud Voice Gateway
G.711µ PCM
DSP
G.711µ
G.729
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
MGCP, H.248,
H.323 SIP
Megaco
• 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
• ITU: www.itu.org
• IETF: www.ietf.org
• SIP: www.cs.columbia.edu/~hgs/sip/
• H.323: www.packetizer.com/iptel/h323/
• MGCP:www.softswitch.org/asp/techlibrary
_protocol.asp?page=techlibrary