1. Introduction to Internet telephony (VoIP)
Kundan Singh
Internet telephony or Voice over IP involves transport of telephone calls over the Internet. Most of the interest in Internet telephony is motivated by cost savings and ease of integrating new services. In a way, Internet telephony is closer to the Internet services such as and web, than to traditional telephony. It employs a variety of protocols, including RTP (Real-time Transport Protocol) for transport of multimedia data and SIP (Session Initiation Protocol) for signaling, i.e., establishing and controlling sessions. SIP is designed to integrate with other Internet services such as , web, voice mail, instant messaging, presence, multi-party conferencing and multimedia collaboration.
In this presentation, I give a brief tutorial of Internet telephony services using SIP.

2. Inside Session Initiation Protocol (SIP) VoIP services using SIP
Agenda
Introducing VoIP
What does it take to build simple audio telephony?
Inside Session Initiation Protocol (SIP)
What are the main features of SIP
VoIP services using SIP
How do we implement various services like call transfer, auto-attendant, voic ?
Introduction to VoIP

3. Audio Packet Transfer …
Digitization (e.g., sampling at 8kHz, 16 bits per sample, i.e, 128 kb/s or 320 bytes per 20 ms)
Real-time compression/encoding (e.g., G.729A at 8 kb/s)
Transport to remote IP address and port number over UDP (Why not TCP?)
Processing on receiver side is the reverse
Introduction to VoIP

4. Sampling, Quantization, Encoding
+127 +0
-127 …
Encode each quantized sample into 8 bit code word
PCM: 8000 x 8 bits = 64 kb/s
Other techniques (differential coding, linear prediction)
2.4 kb/s to 64 kb/s
Sample at twice the highest voice frequency
2 x 4000=8000 Hz (interval of 125 µsec)
Round off samples to one of 256 levels (introduces noise)
Introduction to VoIP

5. Problems with UDP 1 2 3 5 7 6 4 Unreliable UDP Packet loss
timeline
Sender
Receiver
(a)
(b)
Why is TCP not used? Reliability property of TCP requires acknowledgement, which introduces unbound delay. This is not desirable for real-time communication.
Unreliable UDP
Packet loss
Out-of-order (very rarely)
Jitter (delay variation)
Introduction to VoIP

6. Receive buffer (playout buffer or circular buffer)
20 ms
packet
microphone
sendto(remote IP:port)
read
speaker
20 ms
packet
write
get
Received
packet
recvfrom()
put
while (true) {
buf = read(au,20ms); //blocks
if (!silence)
sendto(remote, buf); …
buf = get(20 ms);
write(au, buf); }
Tradeoff in buffer size, streaming vs interactive.
Advanced methods to adapt the buffer size to keep it small enough for lowest delay, and large enough to accommodate most (say 98%) of the packets.
playout buffer
while (true) {
buf = recvfrom(...); // blocks
put(buf); }
Introduction to VoIP

7. Receive buffer (playout buffer or circular buffer)
Receive buffer: to absorb jitter
Tradeoff in buffer size
Adaptive delay adjustment
Sequence number: to detect packet loss; Just ignore the loss!
Sender 1 2 3 5 7 6 4 8 9
Tradeoff in buffer size, streaming vs interactive.
Advanced methods to adapt the buffer size to keep it small enough for lowest delay, and large enough to accommodate most (say 98%) of the packets.
Receiver 3 2 1 1 2 3 5 7 6 8 9
Introduction to VoIP

8. Timestamp vs sequence number
Silence suppression
Variable length packets
Sender t1 t2 t3 t4 t5 t6 t7 t8 t9 1 2 3
Silence … 4 5 6 7 1 2 3 4 5 6 7
Receiver
Playout time vs packet loss detection
Introduction to VoIP

9. Real-time Transport Protocol (RTP)
IP header
8 bits bits bits V CC M
Payload type
Sequence number P X
UDP header
Timestamp (proportional to sampling time)
RTP Header
Source identifier (SSrc)
Encoded
Audio
Optional contributors’ list (CSrc)
msg
RTP: media transport
RTCP: QoS feedback
RFC 3550, 3551
sendto(…, msg, …)
recvfrom(…, msg, …)
Introduction to VoIP

10. Session identified using receive IP address + port
RTP-based conference
ssrc=5263
ssrc=7182
:8000
ssrc=2639
Session identified using receive IP address + port
ssrc=9844
Introduction to VoIP

11. RTP-based conference   Mixer Transcoder
Mixer mixes multiple streams, and puts rtp.ssrcs of contributors in the mixed packet as rtp.csrc
Transcoder converts one encoding to another. Typically to accommodate heterogeneous bandwidth links.
-law
Mixer
Transcoder 
-law 
G.729
G.729
-law
-law
Introduction to VoIP

12. Why do we need signaling?
INVITE for a call using µ-law and G.729 at :8000
OK using µ-law at
Bob=>
Where is Alice?
Alice=>
Sam
Henry=>
Bob
Alice
Sam
Locate destination user
Negotiate session parameters
Henry
Introduction to VoIP

13. Session Initiation Protocol (SIP)
Address similar to
Two stage lookup:
DNS: uses naming authority pointer and service records
Database or service logic: within a domain
office.com
Bob
home.com
Alice
columbia.edu
yahoo.com
Jane
$ dig –t naptr columbia.edu
columbia.edu IN NAPTR 1 0 "s" "SIP+D2U" "" _sip._udp.columbia.edu.
columbia.edu IN NAPTR 2 0 "s" "SIP+D2T" "" _sip._tcp.columbia.edu.
$ dig –t srv _sip._udp.columbia.edu
_sip._udp.columbia.edu IN SRV cocoa.cc.columbia.edu.
_sip._udp.columbia.edu IN SRV eclair.cc.columbia.edu.
$ dig –t a cocoa.cc.columbia.edu
cocoa.cc.columbia.edu IN A
Introduction to VoIP

14. SIP message format INVITE sip:alice@home.com SIP/2.0
Request
INVITE SIP/2.0
From: “Bob”
To: “Alice”
Subject: How are you?
...
Response
SIP/ OK
From: “Bob”
To: “Alice”
Subject: How are you?
...
SIP messages look very similar to HTTP/1.1 requests. Text (instead of binary) format. Requests and responses are similar except the first line. The structure of the message has first line, followed by any number of headers, followed by an optional message body.
The request method name is the operation to perform on the request URI. Proxies typically use the first like to route a request.
Introduction to VoIP

15. Session Description Protocol (SDP)
INVITE
I can support -law and G.729
Send me audio at :6780
Bob
Alice
OK; I can support -law
Send me audio at :8000
ACK
To port 8000
RTP
To port 6780
RTP
Introduction to VoIP

16. SDP message format and offer answer
Request
INVITE SIP/2.0
...
v=0
o=bob IN IP
s=SIP call
c=IN IP
t=0 0
m=audio 6780 RTP/AVP 0 8 5
m=video 6790 RTP/AVP 31
Response
SIP/ OK
...
c=IN IP
t=0 0
m=audio 8000 RTP/AVP 0 8
m=video 0 RTP/AVP 31
Introduction to VoIP

17. Inside Session Initiation Protocol (SIP) VoIP services using SIP
Agenda
Introducing VoIP
What does it take to build simple audio telephony?
Inside Session Initiation Protocol (SIP)
What are the main features of SIP
VoIP services using SIP
How do we implement various services like call transfer, auto-attendant, voic ?
Introduction to VoIP

18. SIP is…, SIP is not …
SIP = core protocol for establishing sessions in the Internet (peer-to-peer)
Transports session description information from initiator (caller) to receiver (callee)
Allows change of parameters in mid-session
Terminate session
NOT for distribution of multimedia data
NOT suitable for media gateway control
. . .
SIP applications typically fall in following categories:
setting up voice-over-IP calls
setting up multimedia conferences
event notification => IM and presence
text and general messaging
signaling transport
Introduction to VoIP

19. Addressing Personal mobility: Examples:
“Alice Smith”
tel:
tel:
pc12.columbia.edu
yahoo.com
columbia.edu
Introduction to VoIP

20. SIP message format Very similar to HTTP/1.1 Headers: Syntax: Requests:
Text-based, request-response
Request method operates on the resource/entity identified in URI
Headers:
To, From, Call-ID, CSeq: transaction identification
Via: response traverses the reverse request path
Content-Length, Content-Type: message body information
Syntax:
White-space doesn’t matter except in first line
Lines can be folded
Multi-valued header fields can be combiled as a comma-list
Requests:
INVITE, ACK, BYE, CANCEL: related to call setup and tear down
OPTIONS, INFO, COMET, PRACK, SUBSCRIBE, NOTIFY, PUBLISH, REFER, …
Responses:
Provisional: 100 Trying, 180 Ringing, 183 Session progress, …
Success: 200 OK, 202 Pending, …
Redirection: 301 Moved permanently, 302 Moved temporarily, …
Request failure: 400 Bad Request, 401 Unauthorized, 404 Not found, 480 Not available, 486 Busy here, …
Server failure: 500 Internal server error, 501 Not implemented, …
Global failure: 600 Busy everywhere, 603 Decline, …
Introduction to VoIP

21. Building blocks SIP user agent SIP redirect server SIP stateless proxy
IP phone, PC, conference bridge,…
SIP redirect server
returns new location for requests
SIP stateless proxy
routes call requests
SIP (forking) stateful proxy
SIP registrar
accepts name to address mapping
Location server
maintains name to address mapping
Maintaining state
stateless: each request and each response handled independently
Fast load balancing proxies; robust
(transaction) stateful: remember a whole request/response transaction
Enterprise servers, . . .
call stateful: remember a call from beginning to end
Billing, NAT traversal, . . .
Other entities: outbound proxy, back-to-back user agent (b2bua), application-level-gateway (ALG), …
Typically implemented in a single software or box
Introduction to VoIP

22. Message routing Response follows the reverse request path
Via header in SIP message records the request path
Via: a.home.com
Via: b.example.com
Via: c.yahoo.com
Via: a.home.com
Via: b.example.com
Via: c.yahoo.com
Request routing decision at each hop
Usually direct end-to-end transport after initial request
Forcing request path: Record-route and Route headers.
Request forking: parallel vs sequential (use q-value in Contact)
Caller and callee info: further govern request routing
INVITE
q=1.0
302 moved
486 busy
200 OK
q=0.7
q=0.2
Introduction to VoIP

23. Example call setup (9) ok (8) (10) (13) (6) unavailable (7) (6) (5)
Alice
(8)
@residence.net
(10)
@visiting.com
(6)
unavailable (7)
(6)
(5)
(3) invite
(4) moved
@school.edu
(12) ok
(6)
Bob
(2) moved
(11) cancel
@home.com
(1) invite
@lab.school.edu
@yahoo.com
Introduction to VoIP

24. Transport
SIP can operate on any packet network, reliable or unreliable
UDP: most common
Low state overhead
But small max packet size
TCP:
Use with SSL
Connection setup overhead
Head of line blocking for trunks (use SCTP instead)
Transport reliability
Request retransmissions, exponential back-off interval
INVITE different than other methods
Retransmit INVITE response after provisional response was sent
Introduction to VoIP

25. NAT traversal Problem: Solutions: Smart servers SIP Signaling Media
(open)SER allows detecting nodes behind a NAT
Use application level gateway and media-proxy
SIP Signaling
Symmetric response routing for UDP (rport)
Connection reuse for TCP/TLS (sip-outbound)
Media
STUN: Simple traversal of UDP through NAT
TURN: Traversal using relay NAT
ICE: Interactive connectivity establishment
iptel.org
INVITE
REGISTER
Contact:
. . .
E= :8123
L= :5060
Introduction to VoIP

26. NAT traversal (ICE) Address gathering Negotiation Connectivity check
STUN server
stun01.sipphone.com
R= :7002
example.net
INVITE (offer)
OK (answer)
E= :8123
L= :8000
:6000
:8000
Gather addresses
:6000 (local)
:6000 (external)
:9004 (relay)
Gather addresses (L,E,R)
:8000 (local)
:8123 (external)
:7002 (relay)
Introduction to VoIP

27. Inside Session Initiation Protocol (SIP) VoIP services using SIP
Agenda
Introducing VoIP
What does it take to build simple audio telephony?
Inside Session Initiation Protocol (SIP)
What are the main features of SIP
VoIP services using SIP
How do we implement various services like call transfer, auto-attendant, voic ?
Introduction to VoIP

28. IP telephony services (PSTN)
Call routing services: pre-call, one party
speed dial
call forwarding
“follow me”
call filtering/blocking (in/out)
do not disturb
distinctive ringing
Call handling features
autoanswer
Multi-party features
call waiting
call transfer (blind, consultative)
conference call
call park
call pickup
music on hold
call monitoring
Services modify call behavior. Can be invoked by user, pre-programmed or programmable feature logic. PSTN its typically pre-subscribed or built-in, but not programmable.
IP telephony can’t win by just providing what PSTN has.
IP-telephony can’t win by just providing what PSTN has
Introduction to VoIP

29. IP telephony services (Internet)
Presence-enabled calls
place call only if callee is available
Presence-enabled conferencing
call conference participants when all are online and not busy
IM conference alerts
receive IM when someone joins a conference
Unified messaging
receive , IM alert for new voic s
Anything that deals with media becomes media services. Usually media and signaling services closely interact with each other, e.g., interactive voice response can invite new participants to join a conference.
Programmability of services
Introduction to VoIP

30. Where do the services reside?
Double ringing sound when boss calls…
Enter your authentication PIN for billing…
Use finger for locating user…
B2BUA
Endpoint
Make call when boss is online …
Proxy/registrar
Fits in basic SIP model or defined as extensions
presence, instant messaging, caller preference, callee capabilities, ...
Integration of web, , IM
Endpoint
Forward to office phone during day, and home phone during evening…
Service control on client vs server
Introduction to VoIP

31. Endpoint call control
Language for End System Services (LESS) for endpoint service creation
Direct user interaction, direct media control
Handle converged information, e.g., call, presence,
Example: when buddy is online, make a call
<less name="online_call"
require="generic presence ui">
<notification status="online" priority="0.5">
<address-switch field="origin">
<address
<call />
<alert sound=“ring.au"
text="Calling …" />
</address>
</address-switch>
</notification>
</less>
Introduction to VoIP

32. Network call control Common gateway interface -- SIP-CGI (RFC 3050)
Call processing language -- CPL (XML-based, allows GUI)
SIP servlets (Java)
Priority.pl
SIP_FROM
SIP_TO
stdin
CGI-PROXY-REQUEST
stdout
SIP proxy
Urgent
Low-priority
Voic
Phone
if (defined $ENV{SIP_FROM} &&
$ENV{SIP_FROM} =~ {
foreach $reg (get_regs())
print "CGI-PROXY-REQUEST $reg SIP/2.0\n";
print "Priority: urgent\n\n"; }
SIP-CGI: Programming language independent. Maintains state via an opaque token. For SIP proxies and endpoints: call routing, controlling forking, call rejection, call modification (Priority, Call-Info). RFC Upload via web or REGISTER
Introduction to VoIP

33. Call transfer REFER Blind/ consultation/ attended A B C active call
REFER C
Referred-By: B
INVITE C
Referred-By: B
BYE A
active call
Introduction to VoIP

34. B2BUA and third-party call control
Back-to-back UA
Incoming call triggers outgoing call
Services
Calling card
Anonymizer B
SIP A C
INVITE
OK (SDP1)
ACK
INVITE (SDP1)
OK (SDP2)
ACK
INVITE (SDP2) OK
ACK
Introduction to VoIP

35. Voicemail Various design alternatives Endpoint based Proxy controls
Redirect after 10s
Endpoint based
Various design alternatives
vmail.pl
SIP_FROM
SIP_TO
stdin
CGI-PROXY-REQUEST
stdout
If no response
Proxy controls
accept after 15s
Voic acts like a phone
CFB=call forward busy
Problems in PSTN:
Voice mail system tied to PBX or phone company
Integration of video, fax, whiteboard?
How to integrate with Internet telephony?
How to integrate with , web and other user applications?
Existing solutions
Voice Profile for Internet Messaging (VPIM)
Web-based unified messaging systems with personalized PSTN voice mail number
Issues
Call reclaiming
Deleting voice/video mail
Integration with PSTN phone
Integration with VPIM, IMAP, POP3
Introduction to VoIP

36. Interactive voice response using VoiceXML
Gateway
VXML Browser
PSTN
Voice gateway
Web server
Service logic (CGI, servlet, JSP)
Voice and telephony functions
VoiceXML browser
Internet user
VXML
HTML
Internet
Telephone
IVR platform
Voice and telephony functions
(ASR, TTS, DTMF)
Service logic (application specific)
Introduction to VoIP

37. VoiceXML contd.
<form>
<field name=‘id’>
<prompt>
Your ID, please.
</prompt>
</field>
<block>
<submit next=“url”/>
</block>
</form>
<form action=“url”>
Enter your Id:
<input name=‘id’>
<input type=‘submit’>
</form>
Telephony, speech synthesis or audio output, user input and grammar, program flow, variable and properties, error handling, …
Introduction to VoIP

38. DTMF PSTN D D Internet Telephone SIP/PSTN Internet user gateway Audio
SIP INFO D
RTP/ RFC2833
Introduction to VoIP

39. Interworking with telephone network
Telephone to IP
Gateway knows the SIP server
ENUM – E164 numbering (using DNS)
=> e164.arpa =>
Suitable for relatively “static” contacts
IP to telephone
Static mapping
Gateway information is dynamic:
Overlapping networks
Multiple providers
Load balancing
Telephony routing over IP (TRIP)
Route advertisement
Can be implemented in outbound proxy
Suitable for current hierarchical network
+1 @service.mci.com at 4¢/min
at 1¢/min
free
Telephone
network
Telephone
subscriber
SIP/PSTN
gateway
SIP server
IP endpoint
Translating audio (µ-law/A-law)
Translating signaling (PRI/T1,ISUP)
Overlap signaling
Advanced features in SIP are lost in PSTN
Translating identifiers (phone number)
Determining transition points
Introduction to VoIP

40. Summary Introducing VoIP Inside SIP VoIP services using SIP
Basic audio transfer, why we need RTP and SIP/SDP?…
Inside SIP
Message format, addressing, building blocks, routing, transport, NAT traversal, …
VoIP services using SIP
Types of services, programmability, call transfer, third-party, voic , interactive voice response, telephone interworking, …
Introduction to VoIP

41. VoIP activities
IETF working groups: sip, sipping, mmusic, xcon, p2psip, simple, impp, iptel, enum, ecrit, avt, sigtran, midcom, …
Elsewhere: 3GPP, ITU-T, W3C, Jabber/XSF, ETSI-Tiphon, IMTC, sip-forum, VON, …
Introduction to VoIP

42. References SIP: RFC 3261-3265 http://www.cs.columbia.edu/sip
RTP: RFC 3550, 3551,
My thesis: (part III: Enterprise IP telephony)
Open source SIP server:
Free SIP accounts:
Introduction to VoIP