Embed Size (px)
Citation preview
SIPping from the Open Source Well
Matthew BynumUC Architect
Agenda
● SIP History● Why SIP matters (SIP and DNS)● Inside the SIP spec● Open Source (and one proprietary) SIP options● What the future entails● Q&A
SIP is a protocol for establishing sessions in an IP network.
SIP History
Glory is fleeting, but obscurity is forever. - Napoleon Bonaparte
Setting the Stage
The Internet Engineering Task Force first met in 1986.
“The mission of the IETF is to make the Internet work better by producing high quality, relevant technical documents that influence the way people design, use, and manage the Internet. “
- http://www.ietf.org/about/mission.html
http://tools.ietf.org/html/rfc5000
IETF Meetings
The First IETF Audiocast occurred in 1992
Create 1Descr.: DNS Discussion San FranOrig.: John Doe [email protected]: http://www.com.comStart: 04.04.2001 / 09.30End: 20.04.2001 / 16:30Media: Audio GSM 224.1.6.7/49000Media: Video H.263 224.1.6.8/49100
Disseminate 2SAP/NNTP/HTTPInviteSMTP/SIP
Join 3PC/Telephone
Media 4PC/Telephone
Simple Conference Invitation Protocol
Session Invitation Protocol
CALLCHANGECLOSE
by Henning Schulzrinneby Mark Handley and Eve Schooler
1xx2xx3xx4xx5xx
UDP/SDP TCP/SCIP
SUCCESSUNSUCCESSFULBUSYDECLINE
UNKNOWNFAILEDFORBIDDENRINGING
RINGINGTRYINGREDIRECTALTERNATIVE
NEGOTIATE
Simple Conference Invitation Protocol
Session Invitation Protocol
SCIP/1.0 302 Callee has moved temporarilyLocation: [email protected]: [email protected] [email protected] 1.0User-Agent: coco/1.3From: Christian Zahl <[email protected]>To: Henning Schulzrinne <[email protected]>Call-Id: [email protected]: ceres.fokus.gmd.deExpires: Mon, 02 Oct 1995 18:44:11 GMTRequired: fc99cb08 audio/pcmu; port=3456; transport=RTP;rate=16000; channels=1; pt=97; net=224.2.0.1; ttl=128,audio/gsm; port=3456; transport=RTP; rate=8000; channels=1,audio/lpc; port=3456; transport=RTP; rate=8000; channels=1
SIP/1.0 REQPA=128.16.65.19 16AU=noneID=128.16.65.19/[email protected][email protected]=0o=van 2353644765 2353687637 IN IP4 128.3.4.5s=Mbone Audioi=Discussion of Mbone Engineering [email protected] (Van Jacobsenc=IN IP4 224.2.0.1/127t=0 0m=audio 3456 RTP PCMU
Papa SIP
“Personal Mobility for Multimedia Services in the Internet”
by Henning Schulzrinne, March 1996http://www.cs.columbia.edu/~hgs/papers/Schu9603_Personal.pdf
http://www.cs.columbia.edu/~hgs/
Creator of RTP
The Internet Architect
http://www.cs.ucl.ac.uk/staff/M.Handley/
SIP (RFC 2543, RFC 3261); SDP (RFC 2327; SAP, RFC 2974); Protocol Independent Multicast-Sparse Mode (PIM-SM, RFC 2362), TCP-Friendly Rate Control (TFRC, RFC 3448), Multicast-Scope Zone Announcement Protocol (MZAP, RFC 2776), Multicast Address Allocation (RFC 2908, RFC 2909), TCP Congestion Window Validation ( RFC 2861), Reliable Multicast ( RFC 3451, RFC 3452, RFC 3453, RFC 3048), Datagram Congestion Control Protocol ( RFC 4340, RFC 4336).
Mark HandleyFounder of XORP (www.xorp.org)
Creator of SDP
SIP Drafts http://www.cs.columbia.edu/sip/history.html
Date Draft Name
December 2, 1996 draft-ietf-mmusic-sip-01
March 27, 1997 draft-ietf-mmusic-sip-02
July 31, 1997 draft-ietf-mmusic-sip-03
November 11, 1997 draft-ietf-mmusic-sip-04
May 14, 1998 draft-ietf-mmusic-sip-05
June 17, 1998 draft-ietf-mmusic-sip-06
July 16, 1998 draft-ietf-mmusic-sip-07
August 7, 1998 draft-ietf-mmusic-sip-08
September 18, 1998 draft-ietf-mmusic-sip-09
September 28, 1998 Last call
November 12, 1998 draft-ietf-mmusic-sip-10
December 15, 1998 draft-ietf-mmusic-sip-11
January 16, 1999 draft-ietf-mmusic-sip-12
February 2, 1999 Approved
March 17, 1999 RFC 2543
SIP Today
RFC 3261 (SIP: Session Initiation Protocol)RFC 3263 (Session Initiation Protocol (SIP): Locating SIP Servers)RFC 3264 (An Offer/Answer Model with Session Description Protocol (SDP))RFC 3265 (Session Initiation Protocol (SIP)-Specific Event Notification)RFC 3325 (Private Extensions to SIP for Asserted Identity within Trusted Networks)RFC 3327 (SIP Extension Header Field for Registering Non-Adjacent Contacts)RFC 3581 (An Extension to SIP for Symmetric Response Routing)RFC 3840 (Indicating User Agent Capabilities in SIP)RFC 4320 (Actions Addressing Issues Identified with the Non-INVITE Transaction in SIP)RFC 4474 (Enhancements for Authenticated Identity Management in SIP)GRUU (Obtaining and Using Globally Routable User Agent Identifiers (GRUU) in SIP)OUTBOUND (Managing Client Initiated Connections through SIP)RFC 4566 (Session Description Protocol)SDP-CAP (SDP Capability Negotiation)ICE (Interactive Connectivity Establishment)RFC 3605 (Real Time Control Protocol (RTCP) Attribute in the Session Description Protocol)RFC 4916 (Connected Identity in the Session Initiation Protocol (SIP))RFC 3311 (The SIP UPDATE Method)SIPS-URI (The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP))RFC 3665 (Session Initiation Protocol (SIP) Basic Call Flow Examples)
http://tools.ietf.org/html/rfc5411
Don’t Panic!
A Hitchhiker's Guide to the Session Initiation Protocol
● Q.931 (TDM)● H.323 (IP)
Alternative protocols…
Why SIP is kind of a big deal
It’s all about the decentralization
Internet
linuxcon.com20.20.20.20SIP Proxy
DNS
SIP
DNS
atlanta.comSIP Proxy
Media
[email protected]@atlanta.com
2.Where is the SIP server for linuxcon.com?20.20.20.20 and port 5061
1.Alice places call to [email protected].
3.INVITE is sent to 20.20.20.20 addressed to [email protected]
4.INVITE is forwarded to the user bob, who answers, and the media is established between Alice and Bob.
SIP and DNS (RFC 3263)
● Use DNS SRV records for determining what servers provide SIP services for a domain (internal and external)
sipserver A 10.0.0.1
; SRV’s_sips._tcp IN SRV 50 1 5061 sipserver.yourdomain.com._sip._tcp IN SRV 90 1 5060 sipserver.yourdomain.com._sip._udp IN SRV 100 1 5060 sipserver.yourdomain.com.
; NAPTRIN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.yourdomain.com.IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.yourdomain.com.IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.yourdomain.com.
SIP and DNS (cont.)
● Use ENUM records for determining what URI a full E.164 number should map to
● Politics suck. Screenshot from the ITU website:
; NAPTR for calling +12561234567
$ORIGIN 7.6.5.4.3.2.1.6.5.2.1.e164.arpa.IN NAPTR 100 10 “u" "E2U+sip" “!^.*$!sip:[email protected]!” .
Inside SIP
User Agents
Client Server
TCP or UDP port 5060TLS on port 5061
SIP Methods
METHOD DESCRIPTION
INVITE Session setupACK Acknowledgement of final response to INVITEBYE Session terminationCANCEL Pending session cancellationREGISTER Registration of a user’s URIOPTIONS Query of options and capabilities
INFO Mid-call signaling transportPRACK Provisional response acknowledgementUPDATE Update session informationREFER Transfer user to a URISUBSCRIBE Request notification of an eventNOTIFY Transport of subscribed event notificationMESSAGE Transport of an instant message bodyPUBLISH Upload presence state to a server
SIP Responses
Status Message
100 Trying180 Ringing183 Session Progress200 OK300 Multiple Choices302 Moved Temporarily305 Use Proxy400 Bad Request401 Unauthorized402 Payment Required403 Forbidden404 Not Found500 Internal Server Error501 Not Implemented502 Bad Gateway
CLASS DESCRIPTION
1xx Provisional or Informational
2xx Success3xx Redirection4xx Client Error5xx Server Error6xx Global Failure
SIP Roles
Element Function
Proxy Responsible for routing
Registrar Accepts REGISTER request from endpoints
Redirect Generates 3xx responses
Back to Back User Agent (B2BUA)
Terminates SIP dialogs from UAC and creates new dialog to end destination
Session Border Controller (SBC)
Demarcation between disparate networks
Media Gateway Media translation
SIP Element Examples
SIP Service Provider
SBCProxy
Registrar/B2BUA
Media GatewaySIP
TDM
RedirectClunky Old PBX
Basic Call Flow
INVITE
Phone BPhone A
180 Ringing
200 OK
ACK
Media
BYE
200 OK
Call Flow with Proxy
INVITE
Proxy (Server/Client)Phone (Client) Phone (Server)
INVITE100 Trying
180 Ringing180 Ringing
200 OK200 OK
ACK
Media
BYE
200 OK
Example SIP INVITE
INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhdsMax-Forwards: 70To: Bob <sip:[email protected]>From: Alice <sip:[email protected]>;tag=1928301774Call-ID: [email protected]: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 142v=0o=alice 2890844526 2890844526 IN IP4 linuxcon.coms=SIP Callc=IN IP4 216.81.194.139t=0 0m=audio 32894 RTP/AVP 0 101a=rtpmap: 0 PCMU/8000a=rtpmap: 101 iLBC/8000
Example SIP OK
SIP/2.0 200 OK Via: SIP/2.0/UDP server10.linuxcon.com;branch=z9hG4bKnashds8;received= 216.81.194.139To: Alice <sip:[email protected]>;tag=1928301774From: Bob <sip:[email protected]>;tag=a6c85cf Call-ID: [email protected]: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdp Content-Length: 131 v=0o=bob 7844 125 IN IP4 10.0.0.1s=SIP Callc=IN IP4 10.0.0.1t=0 0m=audio 43588 RTP/AVP 0a=sendrecva=rtpmap: 0 PCMU/8000
Presence
● Real-time indicator of a person’s willingness and availability to communicate
● Blends communication methods together, allows for designating preferred contact method
SIMPLE – Powering Presence in SIP
● Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions
● Uses the SIP methods of PUBLISH, SUBSCRIBE, and NOTIFY, defined in RFC’s 3903, 3265, and 3856
● http://datatracker.ietf.org/wg/simple/
XMPP– Powering Presence via XML
● EXtensible Messaging and Presence Protocol● Uses XML messages and a
Publisher/Subscriber model for messages, defined in RFC’s 6120, 6121, and 6122
● http://datatracker.ietf.org/wg/XMPP/
Which one should I use?
● Externally facing? Both! Run a “dual-stack” for maximum federation flexibility
● Current IETF draft to bring interoperability: http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core-04
Open Source (and one proprietary) SIP and XMPP Server Options
Knowledge without practice is useless. Practice without knowledge is dangerous.
- Confucius
Two main types of SIP solution
● Back-to-Back User Agent (B2BUA)● owns each leg of call as a separate dialog● Stateful● inter-work SIP with other protocols, including TDM and
analog interfaces● More like traditional telephony● Doesn’t scale as well as a Proxy
● Proxy● Relays messages between UACs and other SIP entities● Stateless option● SIP-only (with some exceptions)● some trouble exists with the way endpoints implement
features (like transfers)● Future-ish proof
Asterisk – B2BUA/Media Server
● B2BUA● Provides ACD, Voicemail, and IVR ● Most popular VoIP project in the world● Backed by Digium in Huntsville, AL● Rooted in traditional telephony● Struggles with NAT traversal
FreeSWITCH
● B2BUA● Provides ACD, Voicemail, and IVR ● Used by other projects for its media
processing capabilities● Geared for replacing a PBX● Basis of Baracuda’s CudaTel product
sipXecs
● Composed of sipX (Proxy), FreeSWITCH (media), OpenFire (IM & Presence)
● Backed by eZuce in Andover, MA; but run by SIPfoundry
● Biggest user is Amazon with 5,000 users● Marketed as an open source Unified
Communications solution
Kamailio
● Registrar, Redirect, Proxy● Fork of “SIP Express Router”● Frequently used to “front-end” other SIP
servers● Does NOT handle media
OpenSIPS
● Registrar, Redirect, Proxy● Fork of “SIP Express Router”● Frequently used to “front-end” other SIP
servers● Does NOT handle media
reSIProcate
● Proxy and Location (repro), STUN/TURN (reTurn)
● Founded in 2002● reSIProcate stacks used by commercial
products(through a “BSD-like” license) from Cisco, Avaya, LifeSize, Plantronics, Motorola, Ericsson, and more
STUN and TURN and ICE, oh my!
● NAT traversal for endpoints is…troublesome● Kamailio or OpenSIPS with RTPproxy or
MediaProxy● reSIProcate (repro + reTurn) (STUN and TURN
but no RFC ICE support)
Proprietary: Cisco UC Manager
● B2BUA for all types of SIP calls (trunk and line)
● Cisco’s implementation is 100% standards
compatible SIP…except when it’s not.● SIP extensions for “feature parity”● Leads to two modes of SIP support for
phones, basic and advanced. Basic is no bueno.
Openfire
● XMPP Server● “Shiny”● Backed by Jive Software● Used by sipXecs for XMPP, has plugins galore
ejabberd
● XMPP Server● Config heavy● Efficient and scalable
Open Source SIP Client Options
Product Version Linux Win Mac Android iOS SIP XMPP NAT
Jitsi 2.2 X X X X X TURN
Blink 0.5.0 X X Pro X ICE
Empathy 3.8.4 X X X ICE
Linphone 3.6.0 X X X X (2.0) X (2.0) X ICE
cSipSimple 1.01 X X ICE
sipdroid 3.1 X X STUN
Future of Communications
How does this get me my flying car?
- Me, a child of the 80’s
SIP-based voice is spreading...
P2P SIP
● Decentralized SIP Services● Uses overlay networks and
Distributed Hash Tables● REsource LOcation And
Discovery (RELOAD)● No RFCs, only drafts
C
AB
http://datatracker.ietf.org/wg/p2psip/
WebRTC
● sipml5.org● HTML5 Web-based SIP clients● Enables future B2C, B2B, P2P, and any other
acronym you can think of●
What do we do now?
More Information
gplus.to/mbynum
linkedin.com/in/mattbynum
slideshare.net/mbynum
www.voip-info.org
www.opentelecoms.org
Asterisk, FreeSWITCH, OpenSIPS books
This work is licensed under a Creative Commons Attribution-ShareAlike 3.0 Unported License.
Q&A
Questions?
The End
“Due to technological advances, changes in consumer preference, and market forces, the question is when, not if, POTS service and the PSTN over which it is provided will become obsolete.”
- AT&T response to FCC on PSTN Evolution, Dec 2009
Appendix
Additional Reference Slides
Offer/Answer Model
INVITE w/SDP (offer)
200 OK w/SDP (answer)
INVITE w/o SDP
200 OK w/SDP (offer)
ACK w/SDP (answer)ACK
Early Offer Delayed Offer
REFER (Transfer)
INVITE
Phone BPhone A Phone C
INVITE
200 OK
200 OK
ACK
ACK
Media Session
REFER (Refer-To: C)
202 Accepted
200 OK
Media Session
NOTIFY
200 OK
BYE
PRACK (Provisional Acknowledgement)
INVITE
100 Trying
183 Session Progress
200 OK
ACK
PRACK
200 OK (PRACK)
PRACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.16.13.87:5060
;branch=z9hG4bKC384From: <sip:[email protected]>;tag=1EDC10-2436To: <sip:[email protected]>;tag=85E9C7C8-A4CDate: Fri, 01 Mar 2002 00:33:42 GMTCall-ID: [email protected]: 102 PRACKRAck: 3696 101 INVITEMax-Forwards: 70Content-Length: 0
OPTIONS Ping
OPTIONS sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKC384From: <sip:[email protected]>;tag=1EDC10-2436To: <sip:[email protected]>;tag=85E9C7C8-A4CCall-ID: [email protected]: 100 OPTIONSContact: <sip:[email protected]>Accept: application/sdpMax-Forwards: 70Content-Length: 0
OPTIONS
200 OK
SIMPLE Presence Example
IP PBX
PUBLISHNOTIFY
SUBSCRIBE
SIMPLE Server
On Hook / Off Hook
XMPP Presence Example
IP PBX
Presence StanzaPresence Stanza
XMPP Server
On Hook / Off Hook
<presence xml:lang="en"> <show>on hook</show> </presence>
![Virgin America - LinuxCon 2010 [Compatibility Mode]](https://img.dokumen.tips/doc/110x75/584935bd1a28aba93a8bf31e/virgin-america-linuxcon-2010-compatibility-mode.jpg)
