20051118 voip standards - hsrm2005/11/18 · 20051118 voip standards cisco public end-to-end ip qos...
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Copyright © 2005, Cisco Systems, Inc. All rights reserved. Printed in USA.
20051118 VoIP Standards
1© 2005 Cisco Systems, Inc. All rights reserved. Cisco Public20051118 VoIP
Standards
VoIPStandards and Status
Johannes Krohn ([email protected])
2© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
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Copyright © 2005, Cisco Systems, Inc. All rights reserved. Printed in USA.
20051118 VoIP Standards
3© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
4© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Voice Packetization
DSPG.711
Analog Digital
(Predigitized)
Compressed Packetized
L3 L2
Reverse Process
Packet Transport
ATM or IP
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)
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20051118 VoIP Standards
5© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
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=
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
6© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
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
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20051118 VoIP Standards
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Standards Cisco Public
Differentiated Services Architecture(RFC 2274, RFC 2275)
Ingress Interior Egress
Complex Traffic Classification
and Conditioning (TCB)
Classification/Marking/Policing
Simple PHB on
Traffic Aggregates
Scheduling/Congestion Avoidance
No Perflow StateTraffic Classes Provisioning
IP/DS Domain
8© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
TTLOffsetIDLenToSByte
DataIPDAIP SAFCSProtoVersionLength
Flow Clontrol
DSCP
UnusedIP Precedence
23456 017
Encoding Traffic Classes (DSCP)
• DiffServ field: the IP version four-header ToS octet or the IP version six traffic-class octet when interpreted in conformance with the definition given in RFC2474
• DSCP: the first six bits of the DiffServ field, used to select a PHB (forwarding and queuing method)
IPv4 Packet
IPv4 IP Precedence
DS Field
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20051118 VoIP Standards
9© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Per-Hop Behaviors (PHB)
• Expedited Forwarding (EF)
Building block for low delay/jitter/loss (voice/video)
Served at a certain rate with short/empty queues
• Assured Forwarding (AF)
Designed for data
High probability of delivery if profile is not exceeded
Four classes and three levels of drop precedence
Specific resources (BW, buffer space) allocated to each class at each node
• Best Effort (BE)
10© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
End-to-End IP QoS for Voice Delay Mechanisms
• Delay is a problem for us/humans, not the technology
• G.114 recommends less than 150-ms end-to-end, one-way delay
• Points of congestion are more susceptible to build delay
• Multiple-delay types
Codec-algorithmic, processing, serialization, propagation
EF PHB
Priority Queuing
(LLQ/IP RTP Priority, MDRR)
Priority Queuing (LLQ/IP RTP Priority)
Link Fragmentation and Interleaving (Serialization on Slow Links)
cRTP (Smaller Packets get Through Faster)
SP (DS)
Customer Customer
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20051118 VoIP Standards
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Standards Cisco Public
End-to-End IP QoS for Voice Delay Variation Mechanisms
• Delay variation causes buffer underruns → codec desynchronization
• Occurs due to mixing traffic of different types from different sources
• Different traffic types meet in output buffers
• Points of congestion are more susceptible to introduce delay variation
vvvvv v v vvvvv
Adaptive Buffer
DSP
“Jittered” Stream
SP (DS)
Customer CustomerEF PHB
Link Fragmentation and Interleaving (Small Packets Interleaved)
12© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
End-to-End IP QoS for Voice Packet Loss Mechanisms
• Voice can survive some packet loss—how much depends on codec
• Points of congestion drop packets due to buffer overflow
• Voice stream is not adaptive (UDP), does not react to WRED
SP (DS)
Customer CustomerEF PHB
Queuing (Guaranteed BW)(LLQ/IP RTP Priority, MDRR)
“Protecting” Real-Time (WRED on Data)
Classification + Policing (CB-Policing/CAR)
Call Admission Control/RSVPQueuing (Guaranteed BW with LLQ/IP RTP Priority)Bandwidth Efficiency Mechanisms (cRTP)
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20051118 VoIP Standards
13© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
14© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Protocol Architecture Models
Traditional
SCCP, MGCP
H.323, SIP
Terminal
MGC
Peer
Client
Host
Peer
Server
• Terminals are managed by the switch/host and cannot talk directly to other terminals
• Peer endpoints can place calls without the presence of a call agent, but may consult call agents/proxies for name resolution/redirection
• Client endpoints cannot initiate calls without their call agent, but media streams flow peer to peer
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20051118 VoIP Standards
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Standards Cisco Public
Distributed Call Control: H.323 and SIP
• All signaling messages and dialed digits are interpreted by the protocol stack on the Endpoint/Gateway
Gateway/Endpoint is an “intelligent”device
• Peer-to-peer call setup (dial plan/IP address servers are optional)
If IP Address Server is out of reach, the Endpoint/GW can choose an alternate route
• TDM signaling types supported is a function of the GW protocol stack
• Resilient over IP connectivity failures
• Scalable – distributed CPU power
• New applications deployed where needed w/o affecting rest of the network components (Internet model)
• Distributed configuration
GK
GW
H.323
Endpoint
IP
PSTN
SIP
PS
SIP
Endpoint
H.323 or
SIP GW
Optional signaling to locate IP address of peer
Call setup signaling
Media
IP Address
Servers
16© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Distributed Call Processing
Call Setup11
E.164 Lookup22
Call Setup33
Ringback88E.164 Lookup44
Call Setup55
Alerting77
Call Connect1010
Connect RTP Stream1111
Ring66
IP WAN11
22
33
88
44
55
77
1010
1111
6699
Off Hook99
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20051118 VoIP Standards
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Standards Cisco Public
Centralized Call Control: MGCP and SCCP
• All signaling messages are “back-hauled” to the Call Agent
Gateway/Endpoint is a “dumb” device
• Call Agent arbitrates all call setup
If Call Agent is out of reach, the Endpoint/GW cannot function
• TDM signaling types supported is a function of the GW and Call Agent
• Dependent on IP network connectivity
Requires failover strategies
• Scalable – central Call Agent is a contention point
• New applications deployed requires Call Agent upgrade (CO model)
• Centralized configuration
GW
IP
PSTN
SCCP
SCCP
Endpoint
MGCP GW
Call setup signaling
Media
Central Call
Agent
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Standards Cisco Public
Centralized Call Processing
CiscoCallManager
IP
WAN
Ring Back44
44
Call Setup33
33
E.164 Lookup22
22
Call Setup11
11
Off Hook66
66
Connect RTP Stream77
77
Ring55
55
Centralized Call Processing
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20051118 VoIP Standards
19© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
20© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
IETF Standards Process
• SIP has an RFC number assigned to it [3261], it’s a “Standard” right?
• Let’s examine the IETF Document Process
• 6 Classes of documents– Standards Track
– Best Current Practice (BCP) - 1 shot Standard
– Informational - not intended to Standardized
– Experimental - not sufficiently studied to Standardize
– Historical - not to be coded to anymore
– Obsoleted - revised by another RFC
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20051118 VoIP Standards
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Standards Cisco Public
IETF Standards Process (Cont.)
• Best Current Practice – 3 stage process– Most often references other RFCs and
Standards to put an “Architecture” together
– Or, could state what “shouldn’t be done”
– Doesn’t generally have new feature
descriptions (called “normative text”)
There are 2 Categories of Standards in the IETF:
• Standards Track – 5 stage process
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IETF Standards Process (Cont.)
Standards Track Documents
• 5 Document levels (called a “Status”)
– First: the Individual “Internet Draft”• Can be written by anybody, or asked for by a WG Chair(s),
or can be the result of a “Design Team” collaboration
• ASCII Formatted only
• Valid for up to 6 months (with an electronic cutoff)
• Can be revised/renewed (theoretically indefinitely)
• Must become an Official Working Group Item at some point
• WG Chair submits a version to the WG as WGLC for “consensus”
• WG Chair submits WG consensus ID to IESG for RFC
“acceptance”
• Assigned an RFC number and published
• It’s now a Standard, right?
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20051118 VoIP Standards
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Standards Cisco Public
IETF Standards Process (Cont.)
Standards Track Documents (cont’d)• Proposed Standard (PS)
– Core of this stage is the “Request for Comments” aspect
– 18 months to 2 years here (best case... usually 3-5 yrs)
– “MUST” have two independent interoperable “completely” compliant implementations demonstrated to IESG to become DS
• Draft Standard (DS)– Considered “worthy” for Standardization
• Generally should have a 3rd implementation to become FS
• Standard (FS or just STD)– Only 62 of these exist !
But there are more than 3825 RFCs... How does that work?!
24© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
IETF Standards Process (Cont.)
Which Category are these Protocols/Apps in?
RADIUS
HTTP/1.1
IPv6
DHCP
MIME
BGP-4
BOOTP
RTP
SIP
MGCP
IPv4
ICMP
UDP
TCP
Telnet
FTP
TFTP
SNMPv3
ARP
DNS
PPP
POP3
OSPFv2
RIPv2
RSVP
IPsec
GRE
Diffserv-EF
Diffserv-AF
LDAPv3
TLS
IMAPv4
MPLS
IKE
COPS
CIDR
SDP
L2TP
SMTP
DVMRP
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20051118 VoIP Standards
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Standards Cisco Public
March 2003 -- SNMPv3 to Full STD
IETF Standards Process (Cont.)
Which Category are these Protocols/Apps in?
RADIUS
HTTP/1.1
IPv6
DHCP
MIME
BGP-4
BOOTP
RTP
RSVP
IPsec
GRE
Diffserv-EF
Diffserv-AF
LDAPv3
TLS
IMAPv4
ARP
DNS
PPP
POP3
OSPFv2
RIPv2
MPLS
IKE
COPS
CIDR
SDP
L2TP
SMTP
DVMRP
Standard
SIP
MGCP
IPv4
ICMP
UDP
TCP
Telnet
FTP
TFTP
SNMPv3
26© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Standard Draft Standard
IETF Standards Process (Cont.)
Which Category are these Protocols/Apps in?
RADIUS
HTTP/1.1
IPv6
DHCP
MIME
BGP-4
BOOTP
RTP
ARP
DNS
PPP
POP3
OSPFv2
RIPv2
Proposed Standard*
SIP
IPv4
ICMP
UDP
TCP
Telnet
FTP
TFTP
SNMPv3
RSVP
IPsec
GRE
Diffserv-EF
Diffserv-AF
LDAPv3
TLS
IMAPv4
MPLS
IKE
COPS
CIDR
SDP
L2TP
SMTP
*No Protocol Extensions can progress until ALL normative references have progressed
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20051118 VoIP Standards
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Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
28© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
H.323 Background
• International (ITU) standard for:
Packet-based multimedia communications systems
• Originated from H.320 (multimedia over ISDN)
• ITU recommendation
v1 approved in 1996, v2 in January 1998, v3 in September 1999,
v4 in November 2000, and v5 in July 2003
• Leverages existing standards, i.e., H.320, Q.931, H.450, etc.
• Wide market acceptance
• Facilitates interoperability between vendors
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20051118 VoIP Standards
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Standards Cisco Public
H.323 Components
• ITU Standard
• Terminal—Endpoint
• Gateway (GW)—TDM to IP conversion
• Gatekeeper (GK)—Phone number and name to IP address lookup and CAC bandwidth management
• Directory Gatekeeper (D-GK)—Tiered hierarchy of GKs
• MCU—Multipoint Control Unit to mix audio and replicate video
GK GK
H.323
Terminals
H.320 Terminal
(ISDN)
H.324 Terminal
(POTS)
GW GW
H.323
Terminals
MCU
D-GK
IPZone1 Zone2
PSTN
30© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Scope of H.323 Recommendation
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
System Control
User Interface
Video I/O
Equipment
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20051118 VoIP Standards
31© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
H.323 Endpoint-to-Endpoint Call Setup
• H.225 Messages (Signaling Protocol) is based on Q.931
Connect
Assumes endpoints (clients)
know each other’s IP Addresses
H.323 Gateway H.323 Gateway
Logical Channel Set-Up (RTP/RTCP)
Logical Channel Set-Up (RTP/RTCP)
Media (UDP)
Bearer
Plane
Bearer
Plane
Set-Up H.225
Signaling
(TCP)
Signaling
Plane
Signaling
Plane
H.245
Signaling
(TCP)
Capabilities Exchange
32© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
IPZone1 Zone2
H.323 Call Setup with Gatekeeper (GK)
GK GK
RRQ
RCF
ARQLRQ
RRQ
RCF
LCFACF
ARQ
ACF
H.225.0 Setup
DRQ
DCF
Active CallDRQ
DCF
H.225.0 Connect with H.245 Capabilities
GWGW
Assumes endpoints don’t
know each other’s IP Address
IP address of the terminating device is
returned by GK to the originating
device
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20051118 VoIP Standards
33© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
34© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Background - Media Gateway Control
SGCP(July 1998)
MGCP 0.1(Nov 1998)
• And then it was proposed to the IETF and ITU, and...
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20051118 VoIP Standards
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Standards Cisco Public
IPDC
MDCP
H.GCP
SGCP
MGCP
MxCP?
ITU-T SG 16IETF MEGACO
Background - Between 11/98 and 3/99
36© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Background - Media Gateway Control
SGCP(July 1998)
MGCP 0.1(Nov 1998)
MGCP 1.0RFC 3435
(Jan 2003)
MGCP 1.0RFC 2705
(Oct 1999)ISC
• International SoftSwitch Consortium (ISC) continued with MGCP: development, extensions, Interop testing
H.248 V1(March 2002)
H.248 V2(May 2002) ITU/ (IETF)
H.248 V0(June 2000)
• ITU/IETF H.248 work was initiated at a point when initial MGCP development was fairly complete
• Both Use Session Description Protocol (SDP, RFC 2327) to describe media capabilities - Just as SIP does
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20051118 VoIP Standards
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Standards Cisco Public
MGCP Concept
“MGCP is designed as an internal protocol within a distributed system that appears to the outside as a single VoIP gateway.”
RFC3435
38© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Media Gateway Controller(Call Agent)
MGCP
IP PSTN
ISUP SS7 GW
SS7SS7
PRIPRIPRIPRI
DSLDSL
MGCP—Components
MGC
TMG
RMG
AMG
Trunking Gateway (TMG): interfaces
SS7 bearer channels to IP network
Access Gateway (AMG): interfaces PBX trunks to IP network
Residential Gateway (RMG): interfaces customer POTS
lines to IP network
Virtual Switch
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20051118 VoIP Standards
39© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
MGCP Call Setup
MGMG
Create Connection Reply with SD
NotifyNotify Ack
Create Connection
User Information Exchange
Create Connection
Notify
Notify Ack
Create Connection Reply with SDModify Connection with returned SD
Delete Ack
Delete
Delete Ack
Delete
Release Compl
Release
Release Compl
Release
MGC
40© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
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20051118 VoIP Standards
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Standards Cisco Public
SCCP-Skinny Client Control Protocol
• Client – Server protocol
• used in Cisco Enterprise IPT solution (CallManager)
• available to external developers
• stimulus based
• no intelligence in the endpoints
42© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
AVVID SCCP Client Registration
Station Register Station Register
Station Register Ack or Rej Station Register Ack or Rej
Station Capabilities Request Station Capabilities Request
Station Capabilities Response Station Capabilities Response
Station Button Template Req Station Button Template Req
Station Button Template Res Station Button Template Res
Station Time Date Req Station Time Date Req
Station Define Time Date Station Define Time Date
DHCP Request DHCP Request
DHCP Reply w/TFTP Addr DHCP Reply with /TFTP Addr
TFTP Request for .cnf TFTP Request for .cnf
TFTP Reply with .cnf file TFTP Reply with .cnf file
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20051118 VoIP Standards
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Standards Cisco Public
AVVID SCCP Client Call Connect
Station Off-Hook
Station Keypad Button
Station Off-Hook
Station Call Info
Station Open Receive Channel
Station Display Text
Station Set Lamp (Steady)
Station Start Tone (Inside)
Station Stop ToneStation Keypad Button
Station Keypad Button Station Call Info
Station Set Lamp (Blink)
Station Set Ringer (Inside Ring)
Station Set Ringer Off
Station Set Lamp (Steady)
Station Open Receive Channel
Station Start Tone (Alerting)
Station Stop Tone
Station Call Info
Station Start Media XmissionStation Open Receive Channel Ack
Station Open Receive Channel AckStation Start Media Xmission
User Information Exchange
44© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
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20051118 VoIP Standards
45© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
SIP Components
• IETF Standard (PS)
• SIP Proxy Server (PS)
Registration Server (REG) – Accepts registration requests from UAs
Redirect Server (RED) – Maps SIP request to one or more addresses
Location Server (LOC) – Provides information on a callees locations
• User Agent (UA)
SIP Gateway (SIP-GW)
IP Phones (SIP)
User Agent
2 - INVITE
3 - INVITE
4 - 3XX Redirect
5 - INVITE
6 - 200 OKAY
7 - 200 OKAY
RTP
1 - REGISTER
SIP-GW
LOC
RED
SIP
PS
46© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Factors, that delayed SIP in the Enterprise
• SIP is relatively new, having just been published in 1999
• SIP was and is still evolving, and certain issues still need to be resolved.
• Solutions using SIP were and are generally not full featured, but rather a subset of the standard features required in enterprise networks today.
• Some offerings use SIP as an encapsulation protocol, thereby negating the interoperability and application benefits of SIP. This implementations out there are propriatary.
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20051118 VoIP Standards
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The Reality: SIP is Ready, SIP is Mature
• The Core SIP Specifications and many of its extensions are at an excellent level of maturity (most of what you need is DONE)
• A lot of very hard problems are solved
• This is ready to implement and folks are implementing now
• Many unpublished specifications are basically ready, but waiting for process overhead
48© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Agenda
• Voice transport over IP
• Signalling Protocol Classification
• IETF Overview
• VoIP signalling protocols
H.323
MGCP
SCCP
SIP
• Myths / Facts
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20051118 VoIP Standards
49© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Will SIP solve everything .... ?
• Endpoints of vendor A will operate with basicfunctionality on Vendor B`s „IP-PBX“ and viceversa.
• There will be vendor dependant extensions that will not interoperate with others.
50© 2005 Cisco Systems, Inc. All rights reserved.20051118 VoIP
Standards Cisco Public
Myths and Facts About SIP
Myth
Fact
This is the greatest misperception of SIP today. SIP defines how
user agents and servers communicate, but does not define
how to implement services
By implementing SIP, all components will be interoperable now and
in the future
Distributed, peer-to-peer protocols are superior to client-server protocols
There are advantages and disadvantages to each. In an enterprise voice system,
centralized control can be more beneficial than distributed.
Scaling is possible with different architectures; the largest VoIPnetworks in the world today use
H.323.
SIP is the only protocol that will allow scaling
SIP is expected to be a leading contributor in the development of new services, however, protocols such as XML already provide
value in enterprise IP Communications
SIP is the only Internet protocol that can drive
voice application development in the
future
Partially true. SIP is designed to work above the IP network, however, it relies heavily on
network functions such as DNS, DHCP, etc.
SIP operates completely
independent of the underlying network
SIP is the future and the whole industry is moving toward it.
SIP is growing in popularity on roadmaps, but it’s not widely
deployed. Many issues still need to be solved.
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The great Voice Myth
"The Great Voice Myth" states that there is only one way to build voice networks, and that there should be only one voice protocol for each function in a packet voice network.
IP is the paradigm shift, not SIP
The ability to choose protocols is fundamental to
the value of IP Communications !!!
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VoIP is about services not about protocols
The question that companies must ask is not
"Which protocol is best ?“
but
"Which services do we want to deploy and which VoIPprotocols best support those services?"
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Conclusion
• Customers need vendors that are committed to support open standards within their products, and are actively developing voice strategies that consider interoperability with all VoIP protocols.
• Without this commitment, VoIP systems are in danger of becoming as proprietary as legacy voice systems.
• Customers need products that support multiple protocols. This way, if a company finds that it needs to migrate its systems or add products that support a different protocol, it will not be required to perform upgrades to the network.
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Q and A
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