Internet Telephony
based on SIP
SMU - Dallas
April 28, May 1, 2000
Henry Sinnreich, MCI WorldCom
Alan Johnston, MCI WorldCom
Internet Multimedia
•
•
•
•
•
•
•
Real Time Protocol (RTP) – media packets
Real Time Control Protocol (RTCP) – monitor & report
Session Announcement Protocol (SAP)
Session Description Protocol (SDP)
Session Initiation Protocol (SIP)
Real Time Stream Protocol (RTSP) – play out control
Synchronized Multimedia Integration Language (SMIL) –
mixes audio/video with text and graphics
References: Search keyword at http://www.rfc-editor.org/rfc.html
For SMIL - http://www.w3.org/AudioVideo/
2
Telephony on the Internet
may not be a stand-alone business, but part of IP services
SIP/RTP Media Architecture
Telephone
Gateway
SIP client
CAS, Q.931, SS7
SG
PCM
MGCP
MG
Public IP Backbone
• Goes everywhere
• End-to-end control
• Consistent for all services
• DNS – mobility
• Messaging
SIP
• Web
• Directory
RTP • Security
• QoS
• Media services
• Sessions
• Telephony
• …………
Any other sessions
3
Commercial Grade IP Telephony
Assure baseline PSTN features
Leverage and Commonality of telephony with the Web/Internet
New services (new revenue)
Scalability (Web-like)
Baseline PSTN&PBX features
•
Client & user authentication
•
Accounting assured QoS
•
QoS assured signaling
•
Security assured signaling
•
Hiding of caller ID & location
Better than PSTN features
•
New & fast service creation
•
Internet (rapid) scalability
•
Mobility
•
Dynamic user preferences
•
End-to-end control
•
Service selection
•
Feature control
•
Mid-call control features
•
Pre-call
•
Mid-call
4
Internet End-to-End Control
Elective
Server
No single point of failure
All services enabled by
protocols: From ftp to web
Internet
R
USER
Elective
Server
User has control of all
applications and choice
of servers
R
USER
R
R
“Dumb Network”
Services supported by
interfaces and central
controllers
ITU Intelligent Network Control:
POTS, ISDN, BISDN, FR, ATM, H.323, MEGACO/H.248, GSM
Central
Control
Central
Control
USER
SW
UNI
SW
SW
NNI
User has little
control
Central
Control
SW
SW
NNI
USER
SW
UNI
5
SIP vs. flavors of IPDC, SGSP, MGCP, MEGACO, H.248
(Internet Client-Server vs. Telco Master-Slave Protocols)
CAS, Q.931, SS7
PSTN
PCM
Legend
CG: gateway Controller
MG: Media Gateway
SIP, H.323
GC
MCGP
MG
Internet
RTP
1. IP Telephony Gateway
Absorbs PSTN complexity at the edge of IP
GC
PSTN
MG
MCGP
GC
MG
IP
RG
PSTN
?
MCGP
Internet
TR 303…
2. “Softswitch” a la IN
•
•
•
phone to phone only
PSTN services
single vendor solution
3. Residential GWY
•
•
•
•
breaks e-2-e control model
no services integration
no choice of server and apps
“unequal access” is reinvented
6
IP Communications
Complete integration of all services under full user control
Web-like:
PSTN/PBX-like:
•
Presence
•
POTS
•
Voice and text chat
•
AIN CS-1, CS-2
•
Messaging
•
PBX & Centrex
•
Voice, data, video
User has control of:
•
Multiparty
•
All addressable devices
 Conferencing
•
Caller and called party
 Education
 Games
preferences
Any quality
Better quality than 3.1 kHz
Most yet to be invented
Mixt Internet-PSTN: Click’nConnect, ICW, unified messaging
7
Development of SIP
•
IETF - Internet Engineering Task Force
–
–
MMUSIC - Multiparty Multimedia Session Control
Working Group
SIP developed by Handley, Schulzrinne, Schooler, and
Rosenberg

–
–
•
Submitted as Internet-Draft 7/97
Assigned RFC 2543 in 3/99
Internet Multimedia Conferencing Architecture.
Alternative to ITU’s H.323
–
–
–
H.323 used for IP Telephony since 1994
Problems: No new services, addressing, features
Concerns: scalability, extensibility
8
SIP Philosophy
•
Internet Standard
–
•
Reuse Internet addressing (URLs, DNS, proxies)
–
•
Utilizes rich Internet feature set
Reuse HTTP coding
–
•
IETF - http://www.ietf.org
Text based
Makes no assumptions about underlying protocol:
–
–
TCP, UDP, X.25, frame, ATM, etc.
Support of multicast
9
SIP Clients and Servers - 1
•
•
SIP uses client/server architecture
Elements:
–
–
SIP User Agents (SIP Phones)
SIP Servers (Proxy or Redirect - used to locate SIP
users or to forward messages.)
•
–
SIP Gateways:
•
•
•
•
Can be stateless or stateful
To PSTN for telephony interworking
To H.323 for IP Telephony interworking
Client - originates message
Server - responds to or forwards message
10
SIP Clients and Servers - 2
Logical SIP entities are:
•
•
User Agents
–
User Agent Client (UAC): Initiates SIP requests
–
User Agent Server (UAS): Returns SIP responses
Network Servers
–
Registrar: Accepts REGISTER requests from clients
–
Proxy: Decides next hop and forwards request
–
Redirect: Sends address of next hop back to client
The different network server types may be collocated
11
SIP Addressing
Uses Internet URLs
– Uniform Resource Locators
– Supports both Internet and PSTN addresses
– General form is [email protected]
– To complete a call, needs to be resolved down to
[email protected]
– Examples:
sip:[email protected]
sip:J.T. Kirk <[email protected]>
sip:[email protected];user=phone
sip:[email protected]
sip:[email protected];phone-context=VNET
12
SIP Session Setup Example
SIP
User Agent
Client
INVITE sip:[email protected]
SIP
User Agent
Server
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
sip.uunet.com
13
Proxy Server Example
SIP
User Agent
Client
SIP
Proxy
Server
INVITE sip:[email protected]
SIP
User Agent
Server
INVITE sip:[email protected]
200 OK
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
server.wcom.com
sip.uunet.com
14
Redirect Server Example
SIP
User Agent
Client
SIP
Redirect
Server
SIP
User Agent
Server
REGISTER [email protected]
200 OK
INVITE sip:[email protected]
302 Moved sip:[email protected]
1
ACK
2
C
3
INVITE sip:[email protected]
RS
UAS
180 Ringing
200 OK
ACK
Media Stream
host.wcom.com
server.wcom.com
sip.uunet.com
15
SIP Requests
SIP Requests (Messages) defined as:
– Method SP Request-URI SP SIP-Version CRLF
(SP=Space,
CRLF=Carriage Return and Line Feed)
– Example: INVITE sip:[email protected] SIP/2.0
M e th o d
D e sc rip tio n
IN V ITE
A se ssio n is b e in g re q u e ste d to b e se tu p u sin g a sp e c ifie d m e d ia
ACK
M e ssa g e fro m c lie n t to in d ic a te th a t a su c c e ssfu l re sp o n se to a n IN V IT E h a s b e e n re c e iv e d
O P TIO N S
A Q u e ry to a se rv e r a b o u t its c a p a b ilitie s
BYE
A c a ll is b e in g re le a se d b y e ith e r p a rty
CANCEL
C a n c e ls a n y p e n d in g re q u e sts. U su a lly se n t to a P ro x y S e rv e r to c a n c e l se a rc h e s
R E G I S T E R U se d b y c lie n t to re g iste r a p a rtic u la r a d d re ss w ith th e S IP se rv e r
16
SIP Requests Example
Required Headers (fields):
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
–
–
–
Uniquely
identify
this
session
request
}
Via: Shows route taken by request.
Call-ID: unique identifier generated by client.
CSeq: Command Sequence number
•
•
generated by client
Incremented for each successive request
17
SIP Requests Example
Typical SIP Request:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]
Subject: Where are you these days?
Content-Type: application/sdp
Content-Length: 124
v=0
o=ajohnston 5462346 332134 IN IP4 host.wcom.com
s=Let's Talk
t=0 0
c=IN IP4 10.64.1.1
m=audio 49170 RTP/AVP 0 3
18
SIP Responses
SIP Responses defined as (HTTP-style):
– SIP-Version SP Status-Code SP Reason-Phrase CRLF
(SP=Space, CRLF=Carriage Return and Line Feed)
– Example: SIP/2.0 404 Not Found
– First digit gives Class of response:
D e sc rip tio n
E x a m p le s
1xx
In fo rm a tio n a l – R e q u e st re c e iv e d , c o n tin u in g to
p ro c e ss re q u e st.
1 8 0 R in g in g
1 8 1 C a ll is B e in g F o rw a rd e d
2xx
S u c c e ss – A c tio n w a s su c c e ssfu lly re c e iv e d ,
u n d e rsto o d a n d a c c e p te d .
200 OK
3xx
R e d ire c tio n – F u rth e r a c tio n n e e d s to b e ta k e n in
o rd e r to c o m p le te th e re q u e st.
3 0 0 M u ltip le C h o ic e s
3 0 2 M o v e d T e m p o ra rily
4xx
C lie n t E rro r – R e q u e st c o n ta in s b a d sy n ta x o r c a n n o t
b e fu lfille d a t th is se rv e r.
4 0 1 U n a u th o riz e d
4 0 8 R e q u e st T im e o u t
5xx
S e rv e r E rro r – S e rv e r fa ile d to fu lfill a n a p p a re n tly
v a lid re q u e st.
5 0 3 S e rv ic e U n a v a ila b le
5 0 5 V e rsio n N o t S u p o rte d
6xx
G lo b a l F a ilu re – R e q u e st is in v a lid a t a n y se rv e r.
6 0 0 B u sy E v e ry w h e re
6 0 3 D e c lin e
19
SIP Responses Example
Required Headers:
SIP/2.0 200 OK
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
– Via, From, To, Call-ID, and CSeq
are copied exactly from Request.
– To and From are NOT swapped!
20
SIP Responses Example
Typical SIP Response
(containing SDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP host.wcom.com
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]
Subject: Where are you these days?
Content-Type: application/sdp
Content-Length: 107
v=0
o=picard 124333 67895 IN IP4 uunet.com
s=Engage!
t=0 0
c=IN IP4 11.234.2.1
m=audio 3456 RTP/AVP 0
21
Forking Proxy Example
SIP
User Agent
Client INVITE
SIP
Proxy
Server
sip:[email protected]
SIP
User Agent
Server 1
INVITE
INVITE
100 Trying
404 Not Found
ACK
SIP
User Agent
Server 2
S1
C
S2
Fork
180 Ringing
180 Ringing
200 OK
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
proxy.wcom.com
sip.mci.com
sip.uunet.com
22
SIP Headers - Partial List
Header
D e sc rip tio n
E x a m p le s
A c c e p t: a p p lic a tio n / sd p
A c c e p t: c u rre n c y / d o lla rs
Accept
In d ic a te s a c c e p ta b le fo rm a ts.
A u th o riza tio n
C o n ta in s e n c ry p tio n in fo rm a tio n
C a ll-ID
U se d to u n iq u e ly id e n tify a
p a rtic u la r se ssio n o r re g istra tio n
m e ssa g e s. S h o u ld h a v e
ra n d o m n e ss to e n su re o v e ra ll
g lo b a l u n iq u e n e ss.
C o n ta c t
A lte rn a tiv e S IP U R L fo r m o re
d ire c t m e ssa g e ro u tin g .
C o n te n t-L e n g th
O c te t c o u n t in m e ssa g e b o d y .
C o n te n t-T y p e
C o n te n t ty p e o f m e ssa g e b o d y
C o n te n t-T y p e : a p p lic a tio n / sd p
c : a p p lic a tio n / h .3 2 3
CSeq
Com m and Sequence num ber –
u se d to d istin g u ish d iffe re n t
re q u e sts d u rin g th e sa m e se ssio n .
CSeq:
CSeq:
CSeq:
CSeq:
E n c ry p tio n
E n c ry p tio n in fo rm a tio n .
E x p ire s
U se d to in d ic a te w h e n th e
m e ssa g e c o n te n t is n o lo n g e r
v a lid . C a n b e a n u m b e r o f
se c o n d s o r a d a te a n d tim e .
A u th o riza tio n : p g p in fo …
C a ll-ID : 1 @ m a rs.b ro o k s.n e t
C a ll-ID : Ja n -0 1 -1 9 9 9 -1 5 1 0 i: 3 1 4 1 5 9 2 6 5 3 5 @ u u n e t.c o m
1 @ se rv e r.m c i.c o m
C o n ta c t: W . R ik e r, A c tin g C a p ta in < rik e r@ sta rfle e t.g o v >
C o n ta c t: ro o m 2 0 3 @ h o te l.c o m ; e x p ire s= 3 6 0 0
m : a d m in @ m c i.c o m
C o n te n t-L e n g th : 2 8 5
1 IN V IT E
1 0 0 0 IN V IT E
4325 BYE
1 R E G IS T E R
E n c ry p tio n : p g p in fo …
E x p ire s: 6 0
E x p ire s: T h u , 0 7 Ja n 1 9 9 9 1 7 :0 0 C S T
23
SIP Headers - Continued
From
R eq uired field containing the orig inating SIP
U R L. C an also includ e a d isp lay nam e.
From : D ana Scully < sip :d ana@ skep tics.org >
From : sip :+ 1 -3 1 4 -3 4 2 -7 3 6 0 @ g atew ay.w com .com ;
tag = 1 2 3 4 5 6 7
f: sip : g uest@ 1 9 2 .1 6 8 .1 .1
M ax-Forw ard s
C ount d ecrem ented b y each server
forw ard ing the m essag e. W hen g oes to
zero, server send s a 4 8 3 T oo M any H op s
resp onse.
M ax-Forw ard s: 1 0
Priority
C an sp ecify m essag e p riority
Priority: norm al
Priority: em erg ency
R ecord -R oute
A d d ed to a req uest b y a p roxy that need s to
b e in the p ath of future m essag es.
R ecord R oute: sip .m ci.com
R eq uire
Ind icates op tions necessary for the session.
R eq uire: local.telep hony
R esp onse -K ey
C ontains PG P key for encryp ted resp onse
exp ected .
R esp onse -K ey: p g p info…
R etry-A fter
Ind icates w hen the resou rce m ay b e
availab le. C an b e a num b er of second s or a
d ate and tim e.
R etry-A fter: 3 6 0 0
R etry-A fter: Sat, 0 1 Jan 2 0 0 0 0 0 :0 1 G M T
24
SIP Headers - Continued
R o u te
D e te rm in e s th e ro u te
ta k e n b y a m e ssa g e .
S u b je c t
C a n b e u se d to in d ic a te
n a tu re o f c a ll.
To
R e q u ire d fie ld c o n ta in in g
th e re c ip ie n t S IP U R L . M a y
c o n ta in a d isp la y n a m e .
U n su p p o rte d
L ists fe a tu re s n o t
su p p o rte d b y se rv e r.
V ia
U se d to sh o w th e p a th
ta k e n b y th e re q u e st.
W a rn in g
C o n ta in s a c o d e a n d te x t to
w a rn a b o u t a p ro b le m
R o u te : o rin o c o .b ro o k s.n e t
S u b je c t: M o re a b o u t S IP
s: Y o u ’d b e tte r a n sw e r!
T o : F o x M u ld e r
< sip :m u ld e r@ lo n e g u n m a n .o rg >
T o : sip :1 0 1 0 9 0 0 0 @ o p e ra to r.m c i.c o m ;
ta g = 3 1 4
t:
sip :1 8 0 0 C O L L E C T @ te le c o m .m c i.c o m ;
ta g = 5 2
U n su p p o rte d : tc a p .te le p h o n y
V ia : S IP / 2 .0 / U D P sip .m fs.c o m
V ia : S IP / 2 .0 / T C P u u n e t.c o m
v : S IP / 2 .0 / U D P 1 9 2 .1 6 8 .1 .1
W a rn in g : 3 3 1 U n ic a st n o t a v a ila b le
25
Via Headers and Routing
•
•
Via headers are used for routing SIP
messages
Requests
–
–
•
Request initiator puts address in Via header
Servers check Via with sender’s address, then add
own address, then forward. (if different, add
“received” parameter)
Responses
–
–
Response initiator copies request Via headers.
Servers check Via with own address, then forward
to next Via address
26
SIP Firewall Considerations
•
Firewall Problem
–
–
•
•
TCP can be used instead of UDP
Record-Route can be used:
–
•
Can block SIP packets
Can change IP addresses of packets
ensures Firewall proxy stays in path
A Firewall proxy adds Record-Route header
–
Clients and Servers copy Record-Route and put in
Route header for all messages
27
SIP Message Body
•
•
Message body can be any protocol
Most implementations:
–
–
SDP - Session Description Protocol
RFC 2327 4/98 by Handley and Jacobson
•
–
–
–
http://www.ietf.org/rfc/rfc2327.txt
Used to specify info about a multi-media session.
SDP fields have a required order
For RTP - Real Time Protocol Sessions:
•
RTP Audio/Video Profile (RTP/AVP) payload descriptions
are often used
28
SDP Examples
SDP Example 1
v=0
o=ajohnston +1-613-555-1212 IN IP4
host.wcom.com
s=Let's Talk
F ie ld
t=0 0
V e rsio n
c=IN IP4 101.64.4.1
m=audio 49170 RTP/AVP 0 3
SDP Example 2
v=0
o=picard 124333 67895 IN IP4
uunet.com
s=Engage!
t=0 0
c=IN IP4 101.234.2.1
m=audio 3456 RTP/AVP 0
D e sc rip to n
v= 0
O rig in
o = < u se rn am e > < se ssio n id > < v e rsio n >
< n e tw o rk ty p e > < ad d re ss ty p e > < ad d re ss>
S e ssio n N a m e
s= < se ssio n n am e >
T im e s
t= < start tim e > < sto p tim e >
C o n n e c tio n D a ta
c= < n e tw o rk ty p e > < ad d re ss ty p e >
< co n n e ctio n ad d re ss>
M e d ia
m = < m e d ia> < p o rt> < tran sp o rt> < m e d ia
fo rm at list>
29
Another SDP Example
v=0
o=alan +1-613-1212 IN host.wcom.com
s=SSE University Seminar - SIP
i=Audio, Listen only
u=http://sse.mcit.com/university/
[email protected]
p=+1-329-342-7360
c=IN IP4 10.64.5.246
b=CT:128
t=2876565 2876599
m=audio 3456 RTP/AVP 0 3
a=type:recvonly
30
Authentication & Encryption
•
SIP supports a variety of approaches:
–
–
•
Proxies can require authentication:
–
–
•
end to end encryption
hop by hop encryption
Responds to INVITEs with 407 ProxyAuthentication Required
Client re-INVITEs with Proxy-Authorization
header.
SIP Users can require authentication:
–
–
Responds to INVITEs with 401 Unathorized
Client re-INVITEs with Authorization header
31
SIP Encryption Example
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston <sip:[email protected]>
To: Jean Luc Picard <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 224
Encryption: PGP version=2.6.2, encoding=ascii
[email protected]@
hWjasGdg,ddgg+fdgf_ggEO;ALewAKFeJqAFSeDlkjhasdf
kj!aJsdfasdfKlfghgasdfasdfa|Gsdf>a!sdasdf3w2945
1k45mser?we5y;343.4kfj2ui2S8~&djGO4kP%Hk#(Khuje
fjnjmbm.sd;da’l;12’;123=]aw;erwAo3529ofgk
32
PSTN Features with SIP
Features implemented by SIP Phone
–
–
–
–
Call answering: 200 OK sent
Busy: 483 Busy Here sent
Call rejection: 603 Declined sent
Caller-ID: present in From header
– Hold: a re-INVITE is issued with IP Addr =0.0.0.0
– Selective Call Acceptance: using From,
Priority, and Subject headers
– Camp On: 181 Call Queued responses are
monitored until 200 OK is sent by the called party
– Call Waiting: Receiving alerts during a call
33
PSTN Features with SIP
Features implemented by SIP Server
– Call Forwarding: server issues 301 Moved
Permanently or 302 Moved Temporarily
response with Contact info
– Forward Don’t Answer: server issues 408
Request Timeout response
– Voicemail: server 302 Moved Temporarily
response with Contact of Voicemail Server
– Follow Me Service: Use forking proxy to try
multiple locations at the same time
– Caller-ID blocking - Privacy: Server encrypts From
information
34
SIP User Location Example
SIP supports mobility across networks and devices
Q=quality gives preference
SIP/2.0 302 Moved temporarily
Contact: sip:[email protected]
;service=IP,voice mail
;media=audio ;duplex=full ;q=0.7
Contact : phone: +1-972-555-1212; service=ISDN
;mobility=fixed; language=en,es, ;q=0.5
Contact : phone: +1-214-555-1212; service=pager
;mobility=mobile
;duplex=send-only ;media=text; q=0.1; priority=urgent
;description=“For emergency only”
Contact : mailto: [email protected]
35
SIP Mobility Support
4
Mobile
Host
SIP Proxy
Server
Foreign
Network
1 INVITE
SIP Redirect
Server
5
7
2 302 moved temporarily
1
2
Corresponding
Host
Home
Network
3
6
Global: Wire and wireless
No tunneling required
3, 4 INVITE
No change to routing
5, 6 OK
For fast hand-offs use:
7 Data
• Use Cellular IP or
• Use DRCP
36
SIP Mobility
Pre-call mobility
•
•
•
MH can find SIP server
via multicast REGISTER
MH acquires IP address
via DHCP
MH updates home SIP
server
Mid-call mobility
•
•
MH->CH: New INVITE
with Contact and
updated SDP
Re-registers with home
registrar
Need not bother home registrar: Use multi-stage registration
Recovery from disconnects
37
Mobile IP Communications
Mobile IP Requirements
•
Transparency above L2:
Move but keep IP address and
all sessions alive
•
Mobility
–
–
–
•
•
•
Within subnet
Within domain
Global
Evolution of Wireless Mobility
•
Circuit Switched Mobility
based on central INs
•
LAN-MAN:
Cellular IP
•
Wide Area: Mobile IP
•
Universal (any net): SIP
AAA and NAIs
Location privacy
QoS for r.t. communications
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Presence, Instant Messaging and Voice
http://www.ietf.org/internet-drafts/draft-ietf-impp-model-03.txt
39
IP SIP Phones and Adaptors
Are Internet hosts
• Choice of application
• Choice of server
1
• IP appliance
Implementations
• 3Com (2)
2
• Cisco
• Columbia University
• Mediatrix (1)
• Nortel (3)
3
• Pingtel
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SIP Summary
•
SIP is:
–
–
–
–
–
•
Relatively easy to implement
Gaining vendor and carrier acceptance
Very flexible in service creation
Extensible and scaleable
Appearing in products right now
SIP is not:
–
–
Going to make PSTN interworking easy
Going to solve all IP Telephony issues (QoS)
41
References
Book on “Internetworking Multimedia” by Jon Crowcroft, Mark Handley,
Ian Wakeman, UCL Press, 1999 by Morgan Kaufman (USA) and
Taylor Francis (UK)
RFC 2543: “SIP: Session Initiation Protocol”
ftp://ftp.isi.edu/in-notes/rfc2543.txt
The IETF SIP Working Group home page
http://www.ietf.org/html.charters/sip-charter.html
SIP Home Page
http://www.cs.columbia.edu/~hgs/sip/
Papers on IP Telephony
http://www.cs.columbia.edu/~hgs/sip/papers.html
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Relevant IETF Working Groups
http://ietf.org/html.charters/wg-dir.html
•
•
•
•
•
•
•
•
•
•
•
•
•
Audio/Video Transport (avt) - RTP
Differentiated Services (diffserv) – QoS in backbone
IP Telephony (iptel) – CPL, GW location, TRIP
Integrated Services (intserv) – end-to-end QoS
Media Gateway Control (megaco) – IP telephony gateways
Multiparty Multimedia Session Control (mmusic) – SIP, SDP,
conferencing
PSTN and Internet Internetworking (pint) – mixt services
Resource Reservation Setup Protocol (rsvp)
Service in the PSTN/IN Requesting InTernet Service (spirits)
Session Initiation Protocol (sip) – signaling for call setup
Signaling Transport (sigtran) – PSTN signaling over IP
Telephone Number Mapping (enum) – surprises !
Instant Messaging and Presence Protocol (impp)
This large work effort may cause the complete re-engineering
of communication systems and services
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SIP Tutorial - Southern Methodist University