What is Internet Telephony?

What is Internet Telephony? Kundan Singh Dec 2010 http://kundansingh.com kundan10@gmail.com This presentation is a subset of http://www.slideshare.net/kundan10/sip-for-geeks

Who am I? PhD from Columbia University in 2006 on reliable, scalable & interoperable Internet telephony Student of Prof. Henning Schulzrinne who is co-inventor of SIP, RTP, RTSP Worked at Adobe with Dr. Henry Sinnreich who is considered God Father of SIP Worked on SIP systems since 1999, building prototypes, systems. Worked at Tokbox and 6Connex on web-based video communication Worked on open source projects p2p-sip, videocity, flash-videoio, restlite, rtmplite, siprtmp VoIP researcher Software architect Software engineer

What will you learn? Voice Sampling Encoding Transport Signaling Call flow Extensions Scripting Challenges On Web NAT traversal Walled garden What is the big picture? How does everything fit together?

… 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) • Transportto remote IP address and port number over UDP (Why not TCP?) • Processing on receiver side is the reverse

+127 +0 -127 Sampling, Quantization, Encoding 10101111…01101101 Encode each quantized sample into 8 bit code word PCM: 8000 x 8 bits = 64 kb/s Others: 2.4 to 64 kb/s What is narrowband/wideband? What are frame vs sample-based codecs? What is Nyquist’s rate? Why 8000 Hz sampling? Round off samples to one of 256 levels (introduces noise)

Signal Processing Techniques • What is lossless vs lossy? • What is speech vs full-frequency encoding? • What are Fourier Transform and MDCT? • What is differential coding? • What is linear prediction? • What is algorithmic delay? What is frame size? • What are CBR, ABR, VBR? • What are filtering, noise cancellation and AGC? • What are silence suppression and VAD? • What are echo suppression and reverb removal? • What is comfort noise? • What is forward error correction?

Voice codecs

timeline Sender 1 2 3 4 5 6 7 (a) (b) 1 2 3 5 7 6 Receiver Problems with UDP • Unreliable UDP • Packet loss • Out-of-order (very rarely) • Jitter (delay variation)

read 20 ms packet sendto(remote IP:port) microphone write get 20 ms packet speaker put Received packet recvfrom() Playout buffer while (true) { buf = read(au,20ms); //blocks if (!silence) sendto(remote, buf); … buf = get(20 ms); write(au, buf); } playout buffer while (true) { buf = recvfrom(...); // blocks put(buf); }

8 9 0 1 2 3 4 1 2 3 4 5 6 7 8 9 0 2 3 4 1 2 3 5 7 6 0 2 1 0 9 3 2 2 3 2 1 1 2 1 3 5 7 5 7 6 8 7 9 8 2 Playout buffer • What is the its purpose? • What should be the buffer size? • How to do adaptive delay adjustment? • Why do you need sequence number? Sender Receiver http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.54.9716

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

8 bits 8 bits 16 bits Real-time Transport Protocol (RTP) IP header 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 sendto(…, msg, …) recvfrom(…, msg, …)

RTP FAQ • Who uses RTP? • Is RTP a transport or application protocol? • Is RTP secure? What are SRTP, ZRTP? • Is RTP header a big overhead? • Is RTCP needed for two-party voice calls? http://www.cs.columbia.edu/~hgs/rtp/

INVITE for a call using µ-law and G.729 at 202.16.49.27:8000 OK using µ-law at 128.59.19.194 Bob=>192.1.2.3 Alice=>128.59.19.194 Where is Alice? 128.59.19.194 Sam 154.28.32.112 Henry=>125.33.2.81 Why do you need signaling? Bob 202.16.49.27 Alice 128.59.19.194 • Locate destination user • Negotiate session parameters Sam 154.28.32.112 Henry 125.33.2.81

What will you learn? Voice Sampling Encoding Transport Signaling Call flow Extensions Scripting Challenges On Web NAT traversal Walled garden

HTTP GET, POST, HEAD, PUT, DELETE SIP REGISTER, INVITE, BYE, CANCEL, ACK, OPTIONS, … Text-based protocol Request-response Headers and body SMTP HELO, MAIL, RCPT, DATA, QUIT RTSP SETUP, TEARDOWN, PLAY, RECORD, DESCRIBE, …

yahoo.com alice24@yahoo.com tel:12125551234 alice@columbia.edu 1234@columbia.edu tel:19172223333 Alice.Smith@columbia.edu columbia.edu pc12.columbia.edu Addressing Examples: “Alice Smith” <sip:alice@columbia.edu> sip:alice@128.59.19.194:5070 sip:alice@columbia.edu;user=phone;transport=tcp What is address-of-record (AoR)? What is sips URI? What is tel URI? – RFC 3966

Lookup Two stage lookup similar to email: DNS: to locate server for a domain Database: locate user within a domain office.com Bob home.com Alice 128.59.19.194 columbia.edu yahoo.com 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 1010 5060 cocoa.cc.columbia.edu. _sip._udp.columbia.edu. 3600 IN SRV 1010 5060 eclair.cc.columbia.edu. $ dig –t a cocoa.cc.columbia.edu cocoa.cc.columbia.edu. 3600 IN A 128.59.59.199 How to do failover and load sharing?

INVITE sip:alice@home.com SIP/2.0 Via: SIP/2.0/UDP 202.16.49.27:5060;branch=z9hG4bK74bf9 Max-Forwards: 70 From: “Bob” <bob@office.com>;tag=9fxced76s1 To: “Alice” <alice@home.com> Call-ID: 384827@202.16.49.27 CSeq: 1 INVITE Contact: <sip:bob@202.16.49.27> Subject: How are you? Content-Type: application/sdp Content-Length: 151 ... Request Response SIP/2.0 200 OK Via: SIP/2.0/UDP 202.16.49.27:5060;branch=z9hG4bK74bf9 From: “Bob” <bob@office.com>;tag=9fxced76s1 To: “Alice” <alice@home.com>;tag=8321234356 Call-ID: 384827@202.16.49.27 CSeq: 1 INVITE Contact: <sip:alice@128.59.19.194> Content-Type: application/sdp Content-Length: 147 ... What are WS and CRLF?

Requests

Responses

Header examples

SIP trapezoid INVITE INVITE INVITE 100 Trying 100 Trying 180 Ringing 180 Ringing 180 Ringing 200 OK 200 OK 200 OK ACK BYE 200 OK Example call flow Proxy Proxy UA UA office.com home.com alice@home.com bob@office.com RTP media session What is an outbound proxy?

Response follows the reverse request path Via header in SIP message records the request path 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 Via: c.yahoo.com Via: b.example.com Via: a.home.com Via: b.example.com Via: a.home.com Via: a.home.com Via: c.yahoo.com Via: b.example.com Via: a.home.com Via: a.home.com Via: b.example.com Via: a.home.com alice@home.com bob@example.com bob@yahoo.com bob@ip2.yahoo.com 302 moved INVITE 486 busy 200 OK Message routing q=1.0 q=0.7 q=0.2

(13) Example call setup (9) ok 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

SIP user agent IP phone, PC, conference bridge,… SIP redirect server returns new location for requests SIP stateless proxy routes call requests SIP (forking) stateful proxy routes call requests 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, . . . Elements Other entities: outbound proxy, back-to-back user agent (b2bua), application-level-gateway (ALG), … Typically implemented in a single software or box

SIP FAQ • What is a SIP transaction? • What is a SIP dialog? • What is early media? • What is re-INVITE? • Why do you need record-route? • What is request forking? http://www.cs.columbia.edu/sip/

RTP To port 8000 RTP To port 6780 Session negotiation INVITE alice@home.com I can support -lawand G.729 Send me audio at 202.16.49.27:6780 Bob Alice 128.59.19.194 202.16.49.27 OK; I can support -law Send me audio at 128.59.19.194:8000 ACK How to modify media session?

Session Description Protocol (SDP) INVITE sip:alice@home.com SIP/2.0 ... Content-Type: application/sdp Content-Length: 151 v=0 o=bob 26172 27162 IN IP4 202.16.49.27 s=- c=IN IP4 202.16.49.27 t=0 0 m=audio6780 RTP/AVP 0 8 5 97 98 a=rtpmap:97 iLBC/8000 a=rtpmap:98 telephone-event/8000 m=video 6790 RTP/AVP 31 What is offer/answer? What is telephone-event?

SIP is…, SIP is not … • Core protocol for establishing sessions • Allows transport of session description • 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 VoIP calls • setting up multimedia conferences • event notification => IM and presence • text and general messaging • signaling transport

What is the real value in SIP? • Open system • Advanced services

Telephony Call routing speed dial, call forwarding, “follow me”, filtering/blocking (in/out), do-not-disturb, distinctive ringing,… Call handling auto-answer, auto-attendant, voice-mail, … Multi-party call waiting, call transfer (blind/consultative), conference call, park, pickup, music-on-hold, monitoring, … Internet Presence enabled place call only if callee is available, invite participants when all are online and not busy, … Unified messaging receive email, IM alert for new voice mail, or when someone joins your conference, … Web enabled click-to-call, web conference, view conference status and voice-mails on web, … what phone has is not enough! Programmable services

Programming services If somebody is calling the third-time, allow mobile. Try office and home in parallel, if that fails, try home. Allow call to mobile if I’ve talked to person before. If on tele-marketing list, forward to dial-a-joke. Try office during day, and mobile in evening. …

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 Forward to office phone during day, and home phone during evening… Endpoint Service control in endpoint vs network?

Endpoint call control • Language for End System Services (LESS) • Direct user interaction, direct media control • Handle converged information, e.g., call, presence, email • Example: call a friend when he comes online • <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 SIP-CGI RFC 3050, CPL, servlets Priority.pl SIP_FROM SIP_TO stdin CGI-PROXY-REQUEST stdout 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"; } } Urgent SIP proxy Phone Low-priority Voicemail

SIP SIP B2BUA and 3pcc B Back-to-back UA • Incoming call triggers outgoing call Services • Calling card • Anonymizer A C INVITE OK (SDP1) ACK INVITE (SDP1) OK (SDP2) ACK INVITE (SDP2) OK ACK

Internet Internet user VXML Voice gateway HTML • Voice and telephony functions • VoiceXML browser Web server Service logic (CGI, servlet, JSP) VoiceXML Gateway VXML Browser PSTN Telephone IVR platform • Voice and telephony functions • (ASR, TTS, DTMF) • Service logic (application specific)

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, …

Interworking with telephone sip:bob@home.com +1-415-123-4567 • 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 Telephone network Telephone subscriber SIP/PSTN gateway SIP server IP endpoint Telephone to IP Gateway knows the SIP server • <sip:4567@gateway2.example.com;user=phone> ENUM – E164 numbering (using DNS) • +1 212 9397042 => 2.4.0.7.9.3.9.2.1.2.1.e164.arpa => sip:hgs@cs.columbia.edu • Suitable for relatively “static” contacts IP to telephone Static mapping • 1-212854xxxx=>@gw1.columbia.edu 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 +1212 @nyc.gw.com at 1¢/min +1212939 @itgw1.columbia.edu free

Summary of services • Call forwarding: basic INVITE behavior • Call transfer: REFER method • Call hold: set media to 0.0.0.0 • Caller id: From, plus extensions • DTMF carriage: carry as RTP (RFC 2833) • Calling card: B2BUA + voice server • Voicemail: UA, proxy, media server • Programming: CGI, CPL, servlet, LESS, CCXML, VoiceXML, SECE, …

What will you learn? Voice Sampling Encoding Transport Signaling Call flow Extensions Scripting Challenges On Web NAT traversal Walled garden

JavaScript API

siprtmp.py in action

HTML Device capture Real-time codecs E2E UDP media Flash Player H.264 Video encoder Echo cancellation UDP media Server socket Hosted & elastic Utility billing Programmable IVR Conferencing Recording/playback Accounting Tracking Phone Voice, SMS, … Browser Cloud

Self management Free resource sharing No central co-ordination … Self management Utility computing Central co-ordination … managed Peer-to-peer ≠ cloud computing

Problem: Solutions: Smart servers 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 INVITE alice@ E=128.59.19.194:8123 REGISTER alice@iptel.org Contact: sip:alice@10.1.2.3:5060 . . . NAT traversal iptel.org L=10.1.2.3:5060

SIP is just a tool!How do you use it? Open house vs walled garden

ITU-T’s H.323 Managed “services” Legacy of telecom network Complex integrated spec IETF’s SIP End-to-end principle Inspired by Internet & web One task for one protocol Winner among VoIP carriers, mobile operators, and digital voice providers Proprietary Adobe’s RTMFP, Skype, Yahoo, MSN, Cisco’s Skinny, … Terminal Control/Devices Jabber/XMPP Jingle for session initiation Re-uses many SIP features Terminal Control/Devices H.245 RTCP Q.931 RAS RTCP SIP SDP Codecs Codecs RTP RTP TPKT TCP UDP Transport Layer IP and lower layers

What do I do? • IP telephony signaling and applications • Technology demonstration by open source • Web communications and peer-to-peer IP telephony

What is Internet Telephony?