sipping from the open source well
TRANSCRIPT
SIPping from the Open Source Well
Matthew BynumUC Architect
A little about me
• Dabbler in Unified Communications for 12 years
• CCIE Voice #21753
• Installed my first Linux distro at age 17 (RedHat 5.0)
• Open Source lover, amateur maker, forestry nerd
Matthew Bynum
http://gplus.to/mbynum
http://www.linkedin.com/in/mattbynum/
Agenda
• SIP History
• Why SIP matters (SIP and DNS)
• Inside the SIP spec
• Open Source (and one proprietary) SIP options
• What the future entails
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
dhcp TCP UDP
TELNET IGMP ICMP FTP ECHO
POP3 OSPF RIP
IETF Meetings
The First IETF Audiocastoccurred in 1992. A method was needed to disseminate the meeting invites.
Create 1Descr.: DNS Discussion San FranOrig.: John Doe [email protected]: http://www.com.comStart: 04.04.2001 / 09.30End: 04.20.2001 / 16:30Media: Audio GSM 224.1.6.7/49000Media: Video H.263 224.1.6.8/49100
Disseminate 2SAP/NNTP/HTTP
InviteSMTP/SIP
Join 3PC/Telephone
Media 4PC/Telephone
Simple Conference
Invitation Protocol
Session Invitation
Protocol
CALL
CHANGE
CLOSE
by Henning Schulzrinneby Mark Handley and Eve Schooler
1xx
2xx
3xx
4xx
5xx
UDP/SDP TCP/SCIP
SUCCESS
UNSUCCESSFUL
BUSY
DECLINE
UNKNOWN
FAILED
FORBIDDEN
RINGINGRINGING
TRYING
REDIRECT
ALTERNATIVE
NEGOTIATE
Simple Conference
Invitation Protocol
Session Invitation
Protocol
SCIP/1.0 302 Callee has moved temporarily
Location: [email protected]
Location: [email protected]
CALL [email protected] 1.0
User-Agent: coco/1.3
From: Christian Zahl <[email protected]>
To: Henning Schulzrinne
Call-Id: [email protected]
berlin.de
Referer: ceres.fokus.gmd.de
Expires: Mon, 02 Oct 1995 18:44:11 GMT
Required: 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 REQ
PA=128.16.65.19 16
AU=none
ID=128.16.65.19/32492374
v=0
o=van 2353644765 2353687637 IN IP4
128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
[email protected] (Van Jacobsen
c=IN IP4 224.2.0.1/127
t=0 0
m=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 (SIP)
• Q.931 (TDM)
• H.323 (IP)
Alternative protocols…
Why SIP is kind of a big deal
It’s all about the decentralization
Internet
linuxcon.com
20.20.20.20
SIP Proxy
DNS
SIP
DNS
atlanta.com
SIP 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
INVITE is sent to
20.20.20.20 addressed
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.
; NAPTR
IN 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 restrict this from being a viable option. 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 5060
TLS on port 5061
SIP Methods
METHOD DESCRIPTION
INVITE Session setup
ACK Acknowledgement of final response to INVITE
BYE Session termination
CANCEL Pending session cancellation
REGISTER Registration of a user’s URI
OPTIONS Query of options and capabilities
INFO Mid-call signaling transport
PRACK Provisional response acknowledgement
UPDATE Update session information
REFER Transfer user to a URI
SUBSCRIBE Request notification of an event
NOTIFY Transport of subscribed event notification
MESSAGE Transport of an instant message body
PUBLISH Upload presence state to a server
SIP Responses
Status Message
100 Trying
180 Ringing
183 Session Progress
200 OK
300 Multiple Choices
302 Moved Temporarily
305 Use Proxy
400 Bad Request
401 Unauthorized
402 Payment Required
403 Forbidden
404 Not Found
500 Internal Server Error
501 Not Implemented
502 Bad Gateway
CLASS DESCRIPTION
1xx Provisional or Informational
2xx Success
3xx Redirection
4xx Client Error
5xx Server Error
6xx 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 BorderController (SBC)
Demarcation between disparate networks
Media Gateway Media translation
SIP Element Examples
Service Provider
SBCProxy
Registrar/B2BUA
Media GatewaySIP
TDM
Redirect
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)
INVITE
100 Trying
180 Ringing
180 Ringing
200 OK
200 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=z9hG4bK776asdhds
Max-Forwards: 70
To: Bob <sip:[email protected]>
From: Alice <sip:[email protected]>;tag=1928301774
Call-ID: [email protected]
CSeq: 314159 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 142
v=0
o=alice 2890844526 2890844526 IN IP4 linuxcon.com
s=SIP Call
c=IN IP4 216.81.194.139
t=0 0
m=audio 32894 RTP/AVP 0 101
a=rtpmap: 0 PCMU/8000
a=rtpmap: 101 iLBC/8000
Example SIP OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP server10.linuxcon.com
;branch=z9hG4bKnashds8;received= 216.81.194.139
To: Bob <sip:[email protected]>;tag=a6c85cf
From: Alice <sip:[email protected]>;tag=1928301774
Call-ID: [email protected]
CSeq: 314159 INVITE
Contact: <sip:[email protected]>
Content-Type: application/sdp Content-Length: 131 v=0
o=alice 7844 125 IN IP4 10.0.0.1
s=SIP Call
c=IN IP4 10.0.0.1
t=0 0
m=audio 43588 RTP/AVP 0
a=sendrecv
a=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 in SIP
• 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/
Open Source (and one proprietary) SIP Server Options
Knowledge without practice is useless. Practice without knowledge is dangerous.
- Confucius
Two main types of SIP servers
• 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
some features (like transfers)– Future proof
Asterisk – B2BUA/Media Server
• B2BUA…so it stays in the signaling (and media) path
• Provides ACD, Voicemail, and IVR functionality
• Most popular VoIP project in the world
• Backed by Digium in Huntsville, AL
• Rooted in traditional telephony
• Struggles with NAT traversal
FreeSWITCH
• B2BUA, stays in the signaling (and media) path
• Provides ACD, Voicemail, and IVR functionality
• Used by other projects for its media processing capabilities
• Geared for replacing a PBX
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
• 1&1 uses Kamailio and has 1 billion minutes per month of usage through the platform
• Frequently used to “front-end” other SIP servers such as Asterisk or FreeSWITCH
• Kamailio does NOT handle media (relies on Asterisk or FreeSWITCH for that)
OpenSIPS
• Registrar, Redirect, Proxy
• Fork of what Kamailio came from (SIP Express Router or SER)
• Frequently used to “front-end” other SIP servers such as Asterisk or FreeSWITCH
• OpenSIPS does NOT handle media (relies on Asterisk or FreeSWITCH for that)
reSIProcate
• Proxy, Location, STUN/TURN
• Initial VOCAL stack started by Vovida Networks “back in the day”, then was acquired by Cisco
• reSIProcate founded in 2002, moved to SIPfoundry, then went independent in 2006
• 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 CallManager (CUCM)
• B2BUA for all types of SIP calls (trunk and line)
• Cisco’s implementation is 100% standards compatible SIP…except when it’s not.
• There are “extensions” to SIP implemented in CUCM for Cisco’s SCCP protocol feature parity to handsets
• Leads to two modes of SIP support for phones, basic and advanced. Basic is no bueno.
Open Source SIP Client Options
Product Version Linux Win Mac Android iOS SIP XMPP NAT Traversal
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
Future of SIP
How does this get me my flying car?
- Me, a child of the 80’s
SIP-based UC 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
•
Where do we go now?
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/2.0
Via: SIP/2.0/UDP 172.16.13.87:5060
;branch=z9hG4bKC384
From: <sip:[email protected]>;tag=1EDC10-2436
To: <sip:[email protected]>;tag=85E9C7C8-A4C
Date: Fri, 01 Mar 2002 00:33:42 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-
CSeq: 102 PRACK
RAck: 3696 101 INVITE
Max-Forwards: 70
Content-Length: 0
OPTIONS Ping
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKC384
From: <sip:[email protected]>;tag=1EDC10-2436
To: <sip:[email protected]>;tag=85E9C7C8-A4C
Call-ID: D110EA36-2BE211D6-801CEF21-
CSeq: 100 OPTIONS
Contact: <sip:[email protected]>
Accept: application/sdp
Max-Forwards: 70
Content-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>