Introduction to VoIP: An SEO-Optimized Title

Introduction to Internet telephony (VoIP)
Kundan Singh
Agenda

Introducing VoIP

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

VoIP services using SIP

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
Sampling, Quantization, Encoding

+127
1010111101101101
Encode each quantized sample into 8 bit code word

+0
PCM: 8000 x 8 bits = 64 kb/s

Other techniques (differential coding, linear prediction) 2.4 kb/s to 64 kb/s
-127
Sample at twice the highest voice frequency 2 x 4000=8000 Hz (interval of 125 sec)
Round off samples to one of 256 levels (introduces noise)

Problems with UDP

timeline Sender
5
(a)
7
(b)
1
Receiver
Unreliable UDP a) Packet loss b) Out-of-order (very rarely) c) Jitter (delay variation)
Receive buffer
(playout buffer or circular buffer)
read microphone 20 ms packet sendto(remote IP:port)
write speaker
20 ms packet
get
put while (true) { buf = read(au,20ms); //blocks if (!silence) sendto(remote, buf); buf = get(20 ms); write(au, buf); } playout buffer
Received packet
recvfrom()
while (true) { buf = recvfrom(...); // blocks put(buf); }
Receive buffer
(playout buffer or circular buffer)

Sender 1 2 3
Receive buffer: to absorb jitter Tradeoff in buffer size Adaptive delay adjustment Sequence number: to detect packet loss; Just ignore the loss!
4 5 6 7 8 9 0 1 2 3 4
1 Receiver 3 2 1
2 1
7 2 1 5 2 3 5 7 6 8 7 9 8 0 9 2 0 3 2
Timestamp vs sequence number

Silence suppression Variable length packets
Sender
t1 1
t2 2
t3 3
t4
t5
t6 4
t7 5
t8 6
t9 7
Silence
1 Receiver
Playout time vs packet loss detection

RealReal-time Transport Protocol (RTP)

IP header UDP header RTP Header Source identifier (SSrc) Optional contributors list (CSrc)
8 bits 8 bits 16 bits
V PX
CC
M Payload type
Sequence number
Timestamp (proportional to sampling time)
msg
Encoded Audio
sendto(, msg, ) recvfrom(, msg, )
RTP: media transport RTCP: QoS feedback
RTPRTP-based conference
ssrc=5263 224.1.2.3:8000
ssrc=7182
ssrc=2639
Session identified using receive IP address + port

ssrc=9844
10
RTPRTP-based conference
Mixer mixes multiple streams, and puts rtp.ssrcs of contributors in the mixed packet as rtp.csrc
Q-law
Transcoder converts one encoding to another. Typically to accommodate heterogeneous bandwidth links.
Mixer
7
Q-law
Q-law
Transcoder
G.729
Q-law
G.729
11
Why do we need signaling?
Bob 202.16.49.27
Alice 128.59.19.194
Sam 154.28.32.112
1. Locate destination user 2. Negotiate session parameters

Henry 125.33.2.81
12
Session Initiation Protocol (SIP)

Address similar to email sip:alice@home.com Two stage lookup:

office.com
Bob
DNS: uses naming authority pointer and service records Database or service logic: within a domain
home.com
columbia.edu yahoo.com Alice 128.59.19.194 Jane 128.59.19.61
$ dig t naptr columbia.edu columbia.edu. 3600 IN NAPTR 1 0 "s" "SIP+D2U" "" _sip._udp.columbia.edu. columbia.edu. 3600 IN NAPTR 2 0 "s" "SIP+D2T" "" _sip._tcp.columbia.edu. $ dig t srv _sip._udp.columbia.edu _sip._udp.columbia.edu. 3600 IN SRV 10 10 5060 cocoa.cc.columbia.edu. _sip._udp.columbia.edu. 3600 IN SRV 10 10 5060 eclair.cc.columbia.edu. $ dig t a cocoa.cc.columbia.edu cocoa.cc.columbia.edu. 3600 IN A 128.59.59.199
13
SIP message format

Request
INVITE sip:alice@home.com SIP/2.0 From: Bob <bob@office.com> To: Alice <alice@home.com> Subject: How are you? ...
Response
SIP/2.0 200 OK From: Bob <bob@office.com> To: Alice <alice@home.com> Subject: How are you? ...
14
Session Description Protocol (SDP)

INVITE alice@home.com I can support Q-law and G.729 Send me audio at 202.16.49.27:6780 OK; I can support Q-law Send me audio at 128.59.19.194:8000 ACK
RTP To port 8000 RTP To port 6780
Bob
Alice
202.16.49.27
128.59.19.194
15
SDP message format and offer answer

Request INVITE sip:alice@home.com SIP/2.0 ... v=0 o=bob 26172 27162 IN IP4 202.16.49.27 s=SIP call c=IN IP4 202.16.49.27 Response t=0 0 m=audio 6780 RTP/AVP 0 8 5 m=video 6790 RTP/AVP 31 SIP/2.0 200 OK ... c=IN IP4 128.59.19.194 t=0 0 m=audio 8000 RTP/AVP 0 8 m=video 0 RTP/AVP 31
Agenda

Introducing VoIP



SIP is , SIP is not

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

Addressing

Personal mobility:
yahoo.com
alice24@yahoo.com alice@columbia.edu 1234@columbia.edu Alice.Smith@columbia.edu

columbia.edu
tel:12125551234
tel:19172223333
pc12.columbia.edu
Examples:

Alice Smith <sip:alice@columbia.edu> sip:alice@128.59.19.194:5070 sip:alice@columbia.edu;user=phone;transport=tcp

SIP message format

Very similar to HTTP/1.1

Requests:

Text-based, request-response Request method operates on the resource/entity identified in URI To, From, Call-ID, CSeq: transaction identification Via: response traverses the reverse request path Content-Length, Content-Type: message body information White-space doesn t matter except in first line Lines can be folded Multi-valued header fields can be combiled as a comma-list
Headers:

INVITE, ACK, BYE, CANCEL: related to call setup and tear down OPTIONS, INFO, COMET, PRACK, SUBSCRIBE, NOTIFY, PUBLISH, REFER, 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,
Responses:

Syntax:

20
Building blocks

SIP user agent

Maintaining state

IP phone, PC, conference bridge, returns new location for requests
SIP redirect server

stateless: each request and each response handled independently

Fast load balancing proxies; robust
SIP stateless proxy

(transaction) stateful: remember a whole request/response transaction

routes call requests

Enterprise servers, . . .
SIP (forking) stateful proxy

routes call requests accepts name to address mapping maintains name to address mapping
call stateful: remember a call from beginning to end

SIP registrar

Billing, NAT traversal, . . .
Location server

Typically implemented in a single software or box
Other entities: outbound proxy, back-to-back user agent (b2bua), application-levelgateway (ALG),
21
Message routing

Response follows the reverse request path

Via header in SIP message records the request path

Via: b.example.com Via: a.home.com Via: b.example.com Via: a.home.com bob@example.com Via: c.yahoo.com Via: b.example.com Via: a.home.com
Via: a.home.com Via: a.home.com alice@home.com

Via: c.yahoo.com Via: b.example.com Via: a.home.com bob@yahoo.com bob@ip2.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 qvalue in Contact) Caller and callee info: further govern request routing
q=1.0
INVITE 486 busy
q=0.7
q=0.2
22
Example call setup

(9) ok
@residence.net
Alice
(10) (6)
(8)
@visiting.com
(13) unavailable (7) (6) (5) (3) invite (12) ok

Bob
@home.com @school.edu
(4) moved (6) (11) cancel

@lab.school.edu
(2) moved (1) invite

@yahoo.com
23
Transport

SIP can operate on any packet network, reliable or unreliable UDP: most common

Transport reliability

Low state overhead But small max packet size Use with SSL Connection setup overhead Head of line blocking for trunks (use SCTP instead)
Request retransmissions, exponential back-off interval INVITE different than other methods

TCP:

Retransmit INVITE response after provisional response was sent
24
NAT traversal

Problem:
iptel.org INVITE alice@
Solutions:

Smart servers

(open)SER allows detecting nodes behind a NAT Use application level gateway and media-proxy Symmetric response routing for UDP (rport) Connection reuse for TCP/TLS (sip-outbound) STUN: Simple traversal of UDP through NAT TURN: Traversal using relay NAT ICE: Interactive connectivity establishment
SIP Signaling

E=128.59.19.194:8123
Media

REGISTER alice@iptel.org Contact: sip:alice@10.1.2.3:5060 ...
L=10.1.2.3:5060
25
NAT traversal (ICE)

STUN server stun01.sipphone.com
R=192.1.2.3:7002
1. 2. 3.
Address gathering Negotiation Connectivity check
INVITE (offer)
example.net OK (answer)
E=128.59.19.194:8123
L=10.1.2.3:8000 10.1.2.3:8000 Gather addresses (L,E,R) 10.1.2.3:8000 (local) 128.59.19.194:8123 (external) 192.1.2.3:7002 (relay)
192.168.1.2:6000
Gather addresses 192.168.1.2:6000 (local) 135.59.22.99:6000 (external) 192.1.2.4:9004 (relay)
26
Agenda

Introducing VoIP



IP telephony services (PSTN)

Call routing services: precall, one party

Multi-party features

speed dial call forwarding follow me call filtering/blocking (in/out) do not disturb distinctive ringing autoanswer

call waiting call transfer (blind, consultative) conference call call park call pickup music on hold call monitoring
Call handling features

IP-telephony can t win by just providing what PSTN has

IP telephony services (Internet)

Presence-enabled calls

place call only if callee is available call conference participants when all are online and not busy receive IM when someone joins a conference receive email, IM alert for new voicemails
Presence-enabled conferencing

IM conference alerts

Unified messaging

Programmability of services
Where do the services reside?

Enter your authentication PIN for billing Use finger for locating user Double ringing sound when boss calls
B2BUA
Make call when boss is online
Endpoint
Proxy/registrar
Forward to office phone during day, and home phone during evening
Endpoint
Service control on client vs server

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, email
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 is=alice@home.com"> <call /> <alert sound=ring.au" text="Calling " /> </address> </address-switch> </notification> </less>
Network call control

Priority.pl
SIP_FROM SIP_TO stdin
CGI-PROXY-REQUEST stdout
Common gateway interface -SIP-CGI (RFC 3050) Call processing language -CPL (XML-based, allows GUI) SIP servlets (Java)
if (defined $ENV{SIP_FROM} && $ENV{SIP_FROM} =~ /sip:boss@mycompany.com/) { foreach $reg (get_regs()) { print "CGI-PROXY-REQUEST $reg SIP/2.0\n"; print "Priority: urgent\n\n"; } } Low-priority Urgent
SIP proxy
Phone
32 Voicemail
Call transfer
A B

REFER Blind/ consultation/ attended
C active call REFER C Referred-By: B INVITE C Referred-By: B BYE A active call
33
B2BUA and third-party call control third
Back-to-back UA

B A
SIP SIP
Incoming call triggers outgoing call Calling card Anonymizer
Services

INVITE OK (SDP1) ACK INVITE (SDP1) OK (SDP2) INVITE (SDP2) OK ACK

ACK
Voicemail
Various design alternatives
Endpoint based
Redirect after 10s
alice@vmail.yahoo.com
Proxy controls
vmail.pl
Voicemail acts like a phone
SIP_FROM SIP_TO stdin
CGI-PROXY-REQUEST stdout
If no response
accept after 15s
35
Interactive voice response

using VoiceXML
Gateway VXML Browser
PSTN
Telephone
Internet
Voice gateway VXML Voice and telephony functions VoiceXML browser
HTML Internet user
IVR platform Voice and telephony functions (ASR, TTS, DTMF) Service logic (application specific)
Web server Service logic (CGI, servlet, JSP)
36
VoiceXML contd.
<form action=url> Enter your Id: <input name=id> <input type=submit> </form> <form> <field name=id> <prompt> Your ID, please. </prompt> </field> <block> <submit next=url/> </block> </form>
Telephony, speech synthesis or audio output, user input and grammar, program flow, variable and properties, error handling,
DTMF
PSTN
Telephone SIP/PSTN gateway
Internet
Internet user Audio
D
SIP INFO
RTP/ RFC2833
38
Interworking with telephone network

+1-415-123-4567 sip:bob@home.com
IP to telephone Telephone to IP Static mapping Telephone Gateway knows the SIP server Telephone SIP/PSTN SIP server
network subscriber gateway <sip:4567@gateway2.example.com ;user=phone>

1-212854xxxx=>@gw1.columbia.edu
IP endpoint
Gateway information is dynamic:
Overlapping networks E164 numbering (using ENUM Translating audio (-law/A-law) Multiple providers DNS) Translating signaling (PRI/T1,ISUP) Load balancing +1 212 9397042 => Overlap signaling 2.4.0.7.9.3.9.2.1.2.1.e164.arp Telephony routing over IP (TRIP) a => sip:hgs@cs.columbia.edu in SIP are Route in PSTN Advanced features lost advertisement

Suitable for relatively static contacts
Can be implemented in outbound proxy Translating identifiers (phone number) hierarchical network Suitable for current +1 Determining transition points @service.mci.com at 4/min +1212 @nyc.gw.com at 1/min +1212939 @itgw1.columbia.edu free

39
Summary

Introducing VoIP

Basic audio transfer, why we need RTP and SIP/SDP? Message format, addressing, building blocks, routing, transport, NAT traversal, Types of services, programmability, call transfer, third-party, voicemail, interactive voice response, telephone interworking,
Inside SIP


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, ETSITiphon, IMTC, sip-forum, VON,
41
References

SIP: RFC 3261-3265 http://www.cs.columbia.edu/sip RTP: RFC 3550, 3551, http://www.cs.columbia.edu/~hgs/rtp My thesis: http://www.cs.columbia.edu/~kns10/publication/thesis.pdf (part III: Enterprise IP telephony) Open source SIP server: http://www.openser.org Free SIP accounts: http://iptel.org
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Introduction to Internet telephony (VoIP)

Inside Session Initiation Protocol (SIP)

VoIP services using SIP

Audio Packet Transfer

Sampling, Quantization, Encoding

Encode each quantized sample into 8 bit code word

PCM: 8000 x 8 bits = 64 kb/s

Round off samples to one of 256 levels (introduces noise)

Problems with UDP

while (true) { buf = recvfrom(...); // blocks put(buf); }

Timestamp vs sequence number

Silence suppression Variable length packets

Playout time vs packet loss detection

RealReal-time Transport Protocol (RTP)

Timestamp (proportional to sampling time)

sendto(, msg, ) recvfrom(, msg, )

RTP: media transport RTCP: QoS feedback

Session identified using receive IP address + port

Why do we need signaling?

1. Locate destination user 2. Negotiate session parameters

Session Initiation Protocol (SIP)

Address similar to email sip:alice@home.com Two stage lookup:

SIP message format

Session Description Protocol (SDP)

SDP message format and offer answer

Inside Session Initiation Protocol (SIP)

VoIP services using SIP

SIP is , SIP is not

alice24@yahoo.com alice@columbia.edu 1234@columbia.edu Alice.Smith@columbia.edu

Alice Smith <sip:alice@columbia.edu> sip:alice@128.59.19.194:5070 sip:alice@columbia.edu;user=phone;transport=tcp

SIP message format

Very similar to HTTP/1.1

SIP user agent

IP phone, PC, conference bridge, returns new location for requests

SIP redirect server

stateless: each request and each response handled independently

Fast load balancing proxies; robust

SIP stateless proxy

(transaction) stateful: remember a whole request/response transaction

routes call requests

SIP (forking) stateful proxy

call stateful: remember a call from beginning to end

Billing, NAT traversal, . . .

Typically implemented in a single software or box

Response follows the reverse request path

Via header in SIP message records the request path

Via: a.home.com Via: a.home.com alice@home.com

Via: c.yahoo.com Via: b.example.com Via: a.home.com bob@yahoo.com bob@ip2.yahoo.com

Request routing decision at each hop

INVITE 486 busy

Example call setup

(13) unavailable (7) (6) (5) (3) invite (12) ok

(4) moved (6) (11) cancel

(2) moved (1) invite

Retransmit INVITE response after provisional response was sent

REGISTER alice@iptel.org Contact: sip:alice@10.1.2.3:5060 ...

NAT traversal (ICE)

Address gathering Negotiation Connectivity check

Gather addresses 192.168.1.2:6000 (local) 135.59.22.99:6000 (external) 192.1.2.4:9004 (relay)

Inside Session Initiation Protocol (SIP)

VoIP services using SIP

IP telephony services (PSTN)

B2BUA and third-party call control third

Suitable for relatively static contacts