SIP: More Than You Ever Wanted To Know About
SIP: More Than You EverWanted To Know AboutJiri Kuthan, TekelecDorgham Sisalem, TekelecMarch 2007All statements are authors’ and may or may not be shared by his company.
Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Outline About This Tutorial Introduction: Why SIP and SIPHistory. Where SIP Was Born: IETFStandardization Introduction to SIP Protocol– SIP Architecture– SIP Servers, ENUM– SIP Message Elements SIP SecuritySIP ServicesBlack-belt SIPSelf-education– Get your hands on SER– Self-test– References IMS BCPs:– QoS– NATs and Firewalls– PSTN RTP – Multimedia ProtocolJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
About This TutorialJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Intended Audience Whoever wishes to gain basic technicalknowledge of SIP protocol: administrators,developers, integrators, CS students. Basic knowledge of TCP/IP networksdesirable. Out-of-scope: detailed developer-level,business aspects.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
About Authors Jiri Kuthan1998: Graduated in CS from U. ofSalzburg, Austria1995-1998: Internship and thesisin Berlin, Germany, FhG Fokus1999-2004: Affiliated asresearcher in Fokus; publishingVoIP publications; involved in theIETF standardization; releasedworld’s leading SIP proxy, SIPExpress Router, with his FhGteam in 20022004: co-founded iptelorg whichwas acquired in 2005 by Tekelec2005: assumed AVP/engineeringposition in Tekelec Dorgham Sisalem1995: Graduated in EE fromTechnical University of Berlin,Germany1995-2005: Researcher and laterdepartment leader whose workresulted in scientific publicationsrelated to QoS, VoIP and securityand being widely quoted byscientific community2000: Obtained PhD from TUBerlin2002: co-founded iptel.org projectwhich later incorporated intoiptelorg2005: assumed position ofDirector Strategic Architecture inTekelecJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Acknowledgments The following persons provided additionalmaterial and/or feedback:––––Raphael Coeffic, Tekelec (QoS)Cristian Constantin, Tekelec (sigcomp)Nils Ohlmeier, TekelecHenning Schulzrinne, Columbia UniversityJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Copyright Notice Authors: Jiri Kuthan, Dorgham Sisalem;Tekelec Copying permitted without explicit authors’permission only if document is not altered.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Introduction: Why SIP?Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
What Is SIP? Depends on WhoYou Are Visionary:Visionary missing piece for running all overIP,IP including your browser, telephone andcoffee machine. Richer user interface thanPSTN. (Quake via DTMF just doesn’t work.)Productivity/collaboration applications. Workfrom anywhere. VP for Business Development:Development technology foralltelephony that allows integrationall-IP-basedIPwith Internet services and surpassinginvestment barriers CFO: reduction of costs by runninghomogenous allall-IP technology. Techie:Techie HTTP-like protocol specified inRFC3261 and associated standards andrunning similarly like Email runs over allall-IP.IPJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
but it is always about ALLALL-IP.PSTNE1DSL Services available to allusers, on-site, off-site,multi-site, underway, homeworking, office-working. Single infrastructure fordata and voice.WaveLAN Effectiveness tools. Service operation can beoutsourced in a Centrex-likeT1manner. Like withweb/email, single servermay host multiple domainsfor better efficiency.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Why to SIP? Challengers:– All-IP nature opens up competition space and removes investmentbarriers. Incumbents: isn’t VoIP a cannibalization threat?– No – it is a Darwinist test in that well-adapting species profit of climechanges whereas the others disappear.– What’s the adaptation chance: running homogenous all-IP networksgreatly reduces cost and increases competitiveness.competitiveness If I was anincumbent, I would pay most attention to key assets: access, identity,retail capability.– Attacking other market segments like challengers do. Why not skypeTM?– That’s a single party game: too few devices because of proprietarytechnology and reportedly the only party to make with skype isskype.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Background Info: Cost-SavingAdvantage in Cost Structure – material provided courtesy of telio.no (2003)All PSTN based providersTelio1002560502525Revenues Consumer’sphone billTerminationcostsPSTNaccess feeContributionmargin350Revenues Consume’sphone billTerminationcostsJiri Kuthan Dorgham Sisalem, Tekelec, March 2007PSTNaccess feeContributionmargin
Pre-IP Telephone Systems Telephonybegan to beknown in 70s ofthe 19’scentury.Inventionauthorshipsubject tocontroversies. Picture:telephoneoperators in1881 in Milan.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
30 Years of VoIP History19771995Late 90s2001200220042005presenceFirst VoIP Publication: D. Cohen, Issues in transnet packetized voicecommunication followed by years of research, mostly QoS-related First Proprietary Solutions Running on the Market. SIP standardizationbegins championed by Prof. H. Schulzrinne.Bitter SIP versus H.323 battles. In ’99, RFC2543 released. End of ’00:SIP chosen for IMS.First sub- -100 SIP telephones/adapters appear.SIP/H.323 Battle is Over – 3GPP R5 released with SIP in it.Standard battlefield moves to presence (Jabber versus Simple). Opensource SIP Express Router is released. RFC 3261 is released.SIP goes to consumers with early adopters: freenet, 1x1, sipphone,telio, Bigger providers roll out: T-mobile/PTT, T-Online-VoIP, BT Broadband, ; PBX deployment base grows too.With deployments in place, manafacturers, operators and standardmakers focused on polishing.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Future of The Phone (1870) Carl Stauber. Waiter: “Our hotel is connectedto all theaters in town. Theopera "Budda" by RichardWagner is just starting in theHoftheater.”(picture found courtesy ofHenry Sinnreich in a Praguehistorical book-shop)Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
2007 – Where Are Thou Really? Construction is Over – Operation Began,Perfection on AgendaPre 20062006 and later Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
SIP Works Fine in 2007 Working standardized technology for runningTelephony over the Internet (and in the future morereal-time applications, such as messaging, gaming,etc.) We have today a variety of interoperable equipment:– clients: hardphone (snom, cisco, mitel, nortel, avaya, .),softphones (microsoft, counterpath, ), dual phones(Nokia), IADs (linksys, AVM), terminal adapters (Sipura)– Gateways: Tekelec, Cisco, Sonus, – Servers: Tekelec/iptelorg, Oracle/hotsip, Ubiquity/Avaya, . Server providers: ISPs (T-Online, Earthlink), ASPs(sipphone, vonage), fixed-mobile-convergenceproviders (telio, truvoice)Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
SIP Gaps in 2007 Operator’s Concerns:– Setting up a network still takes integration effort.– Operation is not yet effort-less either – next product generatingfeaturing automation of common processes, automated securityaudits, and troubleshooting aids yet to come.– Regulatory aspects still moving target.– Reliability sub-nine-fives. (IP availability, NATs, immaturepractices) Visionary Concerns:– New applications still rare.– Security: we shall really not allow being flooded with spam likewith E-mail. Or would you like a 3 AM call from a tele-marketeerfrom other continent? Learner’s Concern: The standards have grown too fat.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Summary of Introduction SIP as of today (2007) is mature all-IPtechnology in polishing stage which is inwide use with an array of equipment fromvarious vendors. Today’s ISP/ASP market is moving tomobile markets. The cost-saving promise holds,applications are coming slowly. Key challenges of adopters: integrationeffort.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
IETF StandardizationJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Where SIP Was Born IETF (www.ietf.org) is a standardization bodywhich has created a large variety of Internetprotocols: TCP/IP for interconnection, SMTP forE-email, FTP for data transfer, RTP for voice,etc. Participation is open: typically folks from bothdata and telecom industry participate, so dofolks from academia. Contribution coming fromindividuals (as opposed to companies). In the past years, the IETF is reinventing itselfand dealing with its increased size and entranceto maintenance mode. SIP-related work grouped in RAI area.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Related IETF Working GroupsCore SpecsSIPPINGNew workfilter to SIPSIPCore signalingprotocol spec.AVTVoice and RTtransportSupporting tocolsENUMConvergence ofIP/PSTNNumberingECRIT IEPREPEmergencyCallsAppsNewcomers and t is it really?)xconSupportingprotocolsSIP P2PTo be definedRetired:Mostly related to PSTN interop (sigtran,Megaco, pint, spirits) and QoS (diffserv, rsvp)Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
‘S’ in SIP Doesn’t Stand forSimpleJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
draft-ietf-sip-hitchhikers-guideFirst Aid: “Hitchhiker’s Guide toGalaxy” SIP WG’s document that aligns all pieces of the SIPpuzzle in a single picture Internet-draft tag: draft-ietf-sip-hitchhikers-guide Refers to the following group of documents: Core SIPSpecifications, PSTN Interworking, General PurposeInfrastructure Extensions, Minor Extensions,Conferencing, , Call Control Primitives, EventFramework and Packages, Quality of Service,Operations and Management, SIP Compression, SIPService URIs, Security Mechanisms , Instant Messagingand Presence, Emergency Services. Read it to get comprehensive view of the “whole picture” Refers to 100 documents! (If you really need to get alist of those that matter, hire me as consultant )Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Introduction to SIPArchitecture-Basic Call Flow-Architectural Fundaments-Protocol PuzzleJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
User Viewpoint: SIP Enddevices User Agent (user application)– UA Client (originates calls)– UA Server (listens for incoming calls) Types of UAs:––––Softphone, hardphones, webphonesMessaging clientsAutomat: PSTN gateways, media servers (voicemail)Etc.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
First Step When Your Phone BootsUpJiri @ 195.37.78.173Location Database#2SIP registrar keeps track ofusers’ whereabouts.This registration exampleestablishes presence ofuser with address jiri@iptel.orgfor one hour and binds thisaddress to user’s current#1 location 195.37.78.173.REGISTER sip:iptel.orgSIP/2.0From: sip:jiri@iptel.orgTo: sip:jiri@iptel.orgContact: sip:195.37.78.173 Expires: 3600#3SIP/2.0 200 OKSIP Registrar(domain iptel.org)Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Basic SIP Call-Flow (ProxySIP Proxy looks up next hops for requestsMode) to served users in location database andforwards the requests there.Location Database#0#6jiriINVITE sip:jiri@iptel.orgFrom:sip:Caller@sip.com;tag 12To: sip: jiri@iptel.orgCall-ID: 345678@sip.com#2#1OK 200From: sip:Caller@sip.com;tag 12To: sip: jiri@iptel.org;tag 34CallCall-ID: 345678@sip.com#7ACK sip:jiri@195.37.78.173jiri@195.37.78.173DNS SRV Query ? iptel.orgReply: IP Address of iptel.org SIP Server#3ProxyINVITE sip:jiri@195.37.78.173From: sip:Caller@sip.com;tag 12To: sip: jiri@iptel.org#4CallCall-ID: 345678@sip.com#5OK 200From: sip:Caller@sip.com;tag 12To: sip: jiri@iptel.org;tag 34CallCall-ID: 345678@sip.comCaller@sip.comMedia streams#8Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007sip:jiri@195.37.78.173
Refresher: IP Design ConceptsRFC1958RFC2775Distributed endend-2-end design*Intelligence and states resides in end-devicesNetwork maintains almost zero intelligence (exceptrouting) and state (except routing tables). End-devices speak to each other using whateverapplications they have. There is almost no logic in thenetwork affecting this behavior. Result: – Flexibility. Introducing new applications is easy.– Failure recovery. No state, no problem on failure.– Scalability. No state, no memory scalability issues.* Manifested in Saltzer-Reed-Clark: End-to-end Arguments in System Design; MIT; 1984Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
draft-rosenberg-sipping-sip-arch(expired)SIP Architecture Borrowsat the Application Layer! To scale well and easily recover from a failure corenetwork infrastructure is kept dumb: SIP serverskeep minimum possible state. Consequently, a greatdeal of intelligence resides in end-devices. (existingexamples: call waiting, video, encryption) Low cost of introduction of new services: SIPservers are largely unaware of the applications: theyset up sessions for audio, video, gaming, whathave-you Make evolution sustainable:sustainable Individual functions areserved by separate protocols: signaling by SIP,media by RTP, interdomain by DNS, etc.Consequently, signaling takes a different path thanmedia!Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Packetized CommunicationSignaling ProtocolMedia TransportEnd UsersCall ServerEnd UsersIP RouterNote: Every packet may take a completely different path Signaling takes typically different path than media does Both signaling and media as well as other applications (FTP,web, email, ) look “alike” up to transport layer and share theJiri Kuthan Dorgham Sisalem, Tekelec, March 2007same fate
AllAll-IP Protocol Zoo (HourglassModel)ENUMiLBC, G.711, .WWWsignaling PSCTPUDPIPv4/IPv6PPPEthernetGPRSSONETAALxV.xJiri Kuthan Dorgham Sisalem, Tekelec, March 2007ATM
SMTP and HTTP/SMTP Legacy SIP is text-oriented protocol –easy to extend and debug The world is split inadministrative domains withDNS names foobar.com,foobar.de, etc. Digest authentication and TLSused for security. Addresses are described usingURIs. Etc.REGISTER sip:iptel.org SIP/2.0Via: SIP/2.0/UDP 212.146.78.122To: sip:bogc@iptel.org CSeq: 671993 REGISTERUser-Agent: Asterisk PBXContact: sip:s@212.146.78.122 ptel.orgJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Protocol Puzzle Session management– Users may move from terminal to terminal with differentcapabilities and change their willingness to communicate– To set-up a communication session between two or more users,a signaling protocol is needed: Session Initiation Protocol (SIP)supports locating users, session negotiation (audio/video/instantmessaging, etc.) and changing session state Media Transport– Getting packetized voice over lossy and congested network inreal-time– RTP – protocol for transmitting real-time data such as audio,video and games End-to-end delivery: underlying IP connects the wholeworldJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Supporting Protocols: How Do I. find domain of called party? Like with email, use DNSto resolve address of server responsible forjiri@iptel.org! authenticate users and generate Call DetailRecords? De-facto RADIUS standard. get over NATs? STUN. More:– set phone clock: NTP– download configuration and firmware: TFTP/FTP/HTTP (nogood standard for usage of these protocols)– resolve phone numbers to SIP addresses? ENUM IETF Practice: Decomposition Principle; Separateprotocols are used for separate purposes. All of them ontop of IP.Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Given All Supporting Protocols are InPlace, What Do I need on SIP Part? SIP Registrar– accept registration requests from users– maintains user’s whereabouts at a Location Server (like GSM HLR) SIP Proxy Server–––––relays call signaling, i.e. acts as both client and serveroperates in a transactional manner, i.e., it keeps no session statetransparent to end-devicesdoes not generate messages on its own (except ACK and CANCEL)Allows for additional services (call forwarding, AAA, forking, etc.) SIP Redirect Server– redirects callers to other servers– Used rather rarely as operators appreciate staying in communicationpath. May be used to achieve very scalable load distribution.All of these elements are logical and are typicallypart of a single server!Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Example SIP NetworkJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Summary of Introduction to SIPArchitecture SIP relies un underlying end-to-end architecture– Most intelligence located in end-devices (similar trend like inmobile industry); networks remains fast-and-simple for betterrobustness– Every simple task in the puzzle is addressed by a single specialpurpose protocol: SIP for signaling, RTP for voice, etc.– Adding a new SIP application does not take change to networkinfrastructure – SIP servers relay any application requests theyreceive Key components of SIP network:––––SIP Phones (User-Agents)SIP Servers (registrar proxy redirect; usually combo)SIP PSTN gatewaysApplications servers (such as media servers)Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
RTP: MultimediaCommunicationJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
IP Based MultimediaCommunication Audio/Video samples are digitized,compressed and sent in UDP packets Compression schemes use limitations ofhuman ears/eyes to reduce bandwidthDiscrete sample signalDigital SignalJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
IP Based MultimediaCommunication Sampled voice is transmitted using RTPprotocol which is separate from SIP SIP establishes the IP addresses and portnumbers at which the end systems cansend and receive data Data packets do not follow the same pathas the SIP packetsJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Real TimeTransport Protocol (RTP)RFC3550 Standardized by the IETF and used by ITU-T as well totransport real-time data such as voice and video Designed to be scalable, flexible and separate data andcontrol mechansms RTP is UDP-based to avoid impaired voice quality whichwould occur if TCP’s flow control hitPHY/MACIPUDPRTPMedia contentPayloadJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
RTP Header Functions Provides information for:–––––––media content typetalk spurtssender identificationsynchronizationloss detectionsegmentation and reassemblysecurity (encryption)Jiri Kuthan Dorgham Sisalem, Tekelec, March 2007
RTP: HeaderV P X M PayloadSequencenumberTimestampSynchronization Source Identifier (SSRC)PayloadJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
RTP Body Can carry multimedia in arbitrary encodingRFC2833: DTMFRFC3016: MPEG-4RFC3385: comfort-noiseRFC3351: basic audio (GSM, G.711, G.729, )and video (H.261, H.263, ) RFC3952: iLBC RFC4298: Broadvoice See http://ietf.org/html.charters/avt-charter.html forallJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Real time Transport ControlProtocol (RTCP) Separate packets sent on a different portnumber Exchange information about losses anddelays between the end systems Packets sent in intervals determinedbased on number of end systems andavailable bandwidth Many implementers don’t bother tosupport RTCPJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Real time Transport ControlProtocol (RTCP) Sender Reports:Reports Information about sentdata, synchronization timestamp Receiver Reports:Reports Information aboutreceived data, losses, jitter and delay Source Description:Name,Email, Phone,DescriptionIdentification Bye:Bye Explicit leave indication Application defined parts:parts Parts forexperimental functionsJiri Kuthan Dorgham Sisalem, Tekelec, March 2007
Audio Quality Largely depends on codec and echocancellation in use Status of the art: codecs with packet lossconcealment su
SIP WG’sdocument that aligns all pieces of the SIP puzzle in a single picture Internet-draft tag: draft-ietf-sip-hitchhikers-guide Refers to the following group of documents: Core SIP Specifications, PSTN Interworking, General Purpose Infrastructure Extensions, Minor Extensions, Conferencing, , Call Control Primitives, Event
SIP SIP phones Blustar 8000i NA SIP SIP phones 9112i, 9133i, 480i Not Supported SIP SIP phones 673xi ( A673xi), 675xi ( A675xi) NA SIP SIP phones 6735i, 6737i ( A6735i, A6737i) NA SIP SIP phones 6739i NA SIP SIP phones 6863i, 6865i, 6867i NA SIP MiVoice Conference phone (UC360
C O N T E N T S Configuration of SIP Trunking for PSTN Access SIP-to-SIP 1 Finding Feature Information 1 Configuration of SIP Trunking for PSTN Access SIP-to-SIP Features 1 Configuring SIP Registration Proxy on Cisco UBE 3 Finding Feature Information 3 Registration Pass-Through Modes 4 End-to-End Mode 4 Peer-to-Peer Mode 5 Registration in Different Registrar Modes 7
How To Guide: SIP Trunking Configuration Using the SIP Trunk Page 6(19) 2.2 The SIP Trunk Page The SIP Trunk pages are found under SIP Trunks. Several SIP Trunk pages may be defined if you have several PBXs or Trunk Services. You need to purchase Additional Trunk Group licensees to get more than one SIP Trunk page. Details are found below. s d he n
How to Guide: SIP Trunking Configuration using the SIP Trunks page 4 2.2 The SIP Trunk Page The SIP Trunk pages are found under SIP Trunks. Several SIP Trunk pages may be defined if you have several PBXs or Trunk Services. You need to purchase Additional Trunk Group licensees to get more than one SIP Trunk page. Details are found below. s d he Tru
To support SIP trunks through a SIP trunk service provider, the SIP Trunk Groups folder was added to the SIP Peers folder in DB Programming. To create a SIP Trunk Group for Fusion Connect Service Provider, navigate to System- Device and Feature Codes- SIP Peers- SIP Trunk Groups and right click in the right hand pane. Then select "Create SIP .
Call Flow Scenarios for Successful Calls This section describes call flows for the following scenarios, which illustrate successful calls: SIP Gateway-to-SIP Gateway—Call Setup and Disconnect, page 7-3 SIP Gateway-to-SIP Gateway—Call via SIP Redirect Server, page 7-6 SIP Gateway-to-SIP Gateway—Call via SIP Proxy Server, page 7-9
4. SIP, VVoIP and QoS 5. SIP and Media Security 6. STIR/SHAKEN and the 'identity' problem 7. Firewalls, NAT and Session Border Controllers 8. SIP Trunking 9. Testing, Troubleshooting and Interoperability 10. ENUM, Peering and Interconnect 11. SIP in the Cloud 12. SIP in Cellular networks 13. SIP and Fax over IP 14. SIP in UC, UCaaS and .
STI-AS IBCF/ TrGW SIP UA Verifier 4. Get Private Key SKS 1. SIP INVITE 22. 200 OK 9. SIP INVITE IBCF/ TrGW CSCF STI-CR CVT 2. SIP INVITE 5. Private Key 7. SIP INVITE (with Identity) 8. SIP INVITE 10. SIP INVITE 11. SIP INVITE 13. Get Certificate 14. Certificate 16. Invoke Analytics 17. Result of Analytics 18. SIP INVITE (with Verification .
