SIP Tutorial

VoIP Workshop Terena 2005 Poznan Poland By Stephen Kingham mailto:[email protected] sip:[email protected]

Stephen Kingham

Copyright Stephen Kingham 2004

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Outline and Objectives

What is SIP Building blocks of a SIP Network SIP Routing Overview of SIP control protocols Overview of SIP media negotiation; SDP Fault finding SIP control protocols SIP Network in AARNet community Why SIP will replace H.323

Stephen [email protected]

SIP Standards
IETF (Internet Standard) RFC 3261 (replaced RFC 2543) Because SIP is an internet standard it leverages and is leveraged by other internet services: Instant Messaging RFC 3428 Presence (The SIMPLE Protocol) Uses SDP, Session Description Protocol RFC 2327, ie what codecs to use. And because SIP is an IETF standard it is designed to fit in with all the other Internet standards. Eg ENUM.

Stephen [email protected]

SIP is PBX/Centrex ready

boss/admin features
call waiting/multiple calls hold transfer RFC 3261 RFC 3264 RFC 3515/Replaces RFC 3261/callee caps message summary package RFC 3261 RFC 3515/Replaces auto attendant call pickup do not disturb call blast Replaces attendant console RFC 3261 night service RFC 3261 RFC 3261 dialog package RFC 3261/2833 simultaneous ringing (forking) basic shared lines barge-in Take Shared-line privacy divert to admin intercom call park RFC 3261 dialog/reg. package Join Replaces dialog package RFC 3261 URI convention

centrex-style features

conference message waiting call forward

attendant features
from Rohan Mahys VON Fall 2003 talk Stephen [email protected]

What is SIP: Use perspective

Heaps simpler than H.323 It is easier to support than H.323 Lots more products than H.323 Cheaper than H.323 Does more than H.323 Has digest authentication (encrypted shared key for users) In practice SIP has more features than H.323. All the Video Manufacturers are moving to SIP! All the PABX manufacturers are moving to SIP! Find a H.323 client for Mac or Unix no you can not! But there are SIP clients for Mac, Unix, PDAs, Microsoft messenger is a Video capable SIP client (support G.722.1 and H.263), GPRS, G3, .!

Lesson: do not do anything with Voice, Video, Telephony, Instant Messaging or Presence unless it uses SIP.
Stephen [email protected]

SIP Forking: Introduction

SIP natively does forking: Make several phones and UAs ring all at the same time. The call is connected to the UA that answers the call. Never need to forward phones to other phones again!!!!

Source RFC 3261

Stephen [email protected]

SIP building blocks

Components: User Agents (UAs) SIP Location Server SIP Redirect Server SIP Proxy Server SIP Back to Back User Agent (b2bua) SIP Gateway

Stephen [email protected]

SIP PROXY Server call flow

DNS
au . du e . rs e nd i l rf o f d r co re .au E u T VI s.ed RV N I r S r 3. inde o f l k @f e o c o ru L b . 2
Flinders University

SIP Proxy Server

SIP Location Server

Call Control Call Control and Audio and Video

err e s stt gii eg R Re 1 1.. u a au u.. E e du ed T TE s.. VII rrs NV e IIN de nd 3 3.. lliin ff @ e@ ce u uc b brr

SIP UA
Stephen [email protected]

SIP UA

SIP REDIRECT Server call flow

DNS
au . du e s. r de n li
Flinders University

Call Control Call Control and Audio and Video

SIP Redirect SIP Location Server Server

2.

ok o L

V R rS o f

u E du.a T I e INV ers. . 3 nd fli @ ce u r b

rd o c re

r fo

ter gis Re 1.

3. INVITE [email protected]

SIP UA

SIP UA

Stephen [email protected]

SIP PROXY Server call flow: from RFC3261

+----+ | UA | | | +----+ | |3)INVITE | [email protected] chicago.com +--------+ V +---------+ 2)Store|Location|4)Query +-----+ |Registrar|=======>| Service|<=======|Proxy|sip.chicago.com +---------+ +--------+=======>+-----+ A 5)Resp | | | | | 1)REGISTER| | | | +----+ | | UA |<-------------------------------+ cube2214a| | 6)INVITE +----+ [email protected] carol

SIP Call flow in more detail (Proxy mode)

UA

DNS

What is SRV record for uni.edu.au

INVITE (with sdp) [email protected] 100 TRYING 180 RINGING 200 OK with sdp ACK

Proxy Server

UA

INVITE (with sdp) [email protected] 180 RINGING 200 OK (after call is answered) with sdp

Media Session, eg Audio and video

Stephen [email protected]

Another view from RFC3261

. atlanta.com proxy . . . biloxi.com proxy . . . Alice's . . . . . . . . . . . . . . . . . . . . Bob's softphone SIP Phone | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | 100 Trying F3 |--------------->| INVITE F4 | |<---------------| 100 Trying F5 |--------------->| | |<-------------- | 180 Ringing F6 | | | 180 Ringing F7 |<---------------| | 180 Ringing F8 |<---------------| 200 OK F9 | |<---------------| 200 OK F10 |<---------------| | 200 OK F11 |<---------------| | |<---------------| | | | ACK F12 | |------------------------------------------------->| | Media Session | |<================================================>| | BYE F13 | |<-------------------------------------------------| | 200 OK F14 | |------------------------------------------------->| Source RFC | 3261 |

SIP Methods
REGISTER - login INVITE start a call ACK CANCEL abort a call setup BYE end a call OPTIONS INFO REFER - Call Transfer MESSAGE - instant messenging SUBSCRIBE / NOTIFY - presence

Source RFC 3261

Response codes used in SIP

1xx: Provisional -- request received, continuing to process the request; 2xx: Success -- the action was successfully received, understood, and accepted; 3xx: Redirection -- further action needs to be taken in order to complete the request; 4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; 5xx: Server Error -- the server failed to fulfil an apparently valid request; 6xx: Global Failure -- the request cannot be fulfilled at any server.
Source RFC 3261

How are the codecs negotiated? Answer SDP

Codecs: the term used to describe the encoding used for the audio and video into data traffic. Eg: Audio G.711 -law and A-law, G.723, G.729, etc Video H.261, H.263, H.264, etc You could have multiple cameras! The sending UA must be able to encode audio/video so that the other end can decode it, and visa versa. sdp is the protocol used by the UAs to tell each other what codecs they support. SDP is embedded into the SIP Messages.
Stephen [email protected]

Presenters note: Review a live call when looking at the following material.

SDP Messages (for reference)

v=0 o=Tesla 289084 289041 IN IP4 lab.high-voltage.org s=c=IN IP4 100.101.102.103 t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 v = Protocol Version number (ignored by SIP) o = <username> <session id> <version> <network type> <address type> <address> (only 3rd field (version) used by SIP) s = Session Name <start time> <stop time> (ignored by SIP) c = <network type> <address type> <connection address> Connection information (IN =internet, IP4 = IPv4, IP Address) t = Time (ignored by SIP) m = Media Name and Transport Address (type, port, RTP/AVP Profile) <media> <port> <transport> <fmt list> a = Attribute (profile, codec, sampling rate)

SDP Messages (the parts that are interesting)

m <media> <port> <transport> <fmt list> Media Name and Transport Address. Define all the audio capabilities, following by a attributes which have details of each a = Attribute (profile, codec, sampling rate). Here is and eg for audio: m=audio 49170 RTP/AVP 0 8 112 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 G7221/16000 There could be a second and third m with attributes for video, data, and/or application. Here is one defining Video supported codecs on this UA. m=video 49916 RTP/AVP 34 31 a=rtpmap:34 H263/90000 a=rtpmap:31 H263/90000
Source RFC 2327

SIP Addressing in the future will be the preferred address, in addition to Telephone numbers

A. G. Bell did not say:

+61-2-6222 3575, come here. I need you!
Ben Teitelbaum @ Internet2

I will prefer to call people using sip:[email protected]

Within the next year you will see this on the bottom of email footers and on business cards of Australian Universities.

SIP.edu Architecture: An achievable goal for a University

SIP User Agent INVITE DNS SRV query (sip:[email protected]) sip.udp.aarnet.edu.au Aarnet.edu.au

DNS

SIP Proxy

SIP-PBX INVITE (sip:[email protected] Gateway

PRI / CAS

PBX

Telephone Number for Stephen Campus Directory

Stephen's Phone
Ref: http://voip.internet2.edu/SIP.edu

AARNet SIP & H.323 network (an example of the building blocks)

SIP Server
Translate telephone numbers to IP addresses

Stephen [email protected]

SIP History

H.323
ITU-T protocol May 1995 Study Group 16 Now V.5

SIP
IETF protocol Became proposed standard in March 1999. Working Groups: SIP, SIPPING, and SIMPLE Now RFC 3261

from Quincy Wus talk, http://www.apan.net Cairns 2004

H323-SIP Comparison of Components

H.323

SIP
SIP UA Registrar, Redirect Server, Proxy Server Conference Server PSTN Gateway

End Station Network Server

Terminal Gatekeeper

MCU PSTN Gateway

from Quincy Wus talk, http://www.apan.net Cairns 2004

H323-SIP Comparison of Protocols

H.323
Signaling Capacity Negotiation Codecs Real-time Communication RAS/Q.931 H.245 Any RTP/RTCP SIP SDP Any

SIP

RTP/RTCP

from Quincy Wus talk, http://www.apan.net Cairns 2004

H323-SIP Comparison of Protocols (cont.)

H.323
Message Encoding Transport Data Conference Instant Message Inter-Domain Routing Annex G Binary UDP and TCP Mostly TCP T.120 ASCII

SIP

UDP and TCP Most UDP RFC 3428 DNS

from Quincy Wus talk, http://www.apan.net Cairns 2004

SIP and DNS

DNS is integral to SIP routing. DNS is used to find a priority list of SIP servers for a domain using in SIP specific SRV records into the DNS. Just like MX records in DNS for mail. So it turns out it is easy to have backup servers in SIP. Good description found on the MIT Internet2 sip.edu project cookbook: http://mit.edu/sip/sip.edu/dns.shtml

Stephen [email protected]

SIP and DNS

Specific SRV records added to your DNS for SIP, eg
IN A 192.94.63.28
;If we place the SRV record above the next line it fails to load

$ORIGIN aarnet.edu.au. _sip._udp SRV 0 1 5060 ser.yarralumla.aarnet.edu.au. _sip._udp SRV 1 1 5060 ser.nsw.aarnet.edu.au. ser.yarrulumla.aarnet..edu.au. IN A 192.94.63.28 ser.nsw.aarnet..edu.au. IN A 138.44.16.90

Stephen [email protected]

SIP and DNS TEST

On a unix host use the dig command: dig -t SRV _sip._udp.aarnet.edu.au You should get a response that has this in it:
;; QUESTION SECTION: ;_sip._udp.aarnet.edu.au. IN SRV

;; ANSWER SECTION: _sip._udp.aarnet.edu.au. 333 IN

SRV 1 1 5060 ser.yarralumla.aarnet.edu.au.

Stephen [email protected]

SIP and ENUM

ENUM (rfc 2916) uses the DNS to find the full SIP address using a telephone number. ACA will have ENUM Tier 1 into Australia on 6 June 2005
http://www.aca.gov.au/telcomm/telephone_numbering/enum_nsg2/.

Stephen [email protected]

2. Today, many addresses

tel:+61 2 6222 3535 mailto:[email protected] tel:+61 2 6222 3575 sip:[email protected]

Source: Patrik Fltstrm, Area Director Applications Area IETF, from ITU Tutorial Workshop on ENUM 8 Feb 2002 Geneva

2. With ENUM, only one

ENUM returns all of these for the caller to choose from: tel:+61 2 6222 3535 mailto:[email protected] tel:+61 2 6222 3575 sip:[email protected]

Hand out enum enabled number +61 2 6222 3575

Source: Patrik Fltstrm, Area Director Applications Area IETF, from ITU Tutorial Workshop on ENUM 8 Feb 2002 Geneva

VoIP via PSTN to PSTN

SIP-Server ENUM SIP-Server Forked SIP call DNS-Server

Gateway Gateway

Adapted from: Patrik Fltstrm, Area Director Applications Area IETF, from ITU Tutorial Workshop on ENUM 8 Feb 2002 Geneva

ENUM in a nutshell
take phone number turn into domain name ask the DNS return list of URIs mailto:[email protected] sip:[email protected]

+46 86859131

1.3.1.9.5.8.6.8.6.4.e164.arpa.

Source: Patrik Fltstrm, Area Director Applications Area IETF, from ITU Tutorial Workshop on ENUM 8 Feb 2002 Geneva

SIP and TRIP (Telephone Routing over IP)

TRIP (rfc 3219 not passed) does for telephone numbers that BGP does for the entire Internet. Dynamic routing by advertisement! More research and experimentation needed here. perhaps a simpler form of TRIP (STRIP?) encapsulated in MIME?
[Source: Discussions between Randy Bush, Andrew Rutherford and Stephen Kingham 3 Feb 2004].

Stephen [email protected]

References used in this talk

Henning Schulzrinne's Marvelous SIP Page http://www.cs.columbia.edu/sip/ Internet2 SIP.edu initiative http://voip.internet2.edu/SIP.edu/ take a look at the CookBook. http://www.iptel.org/ home of The SIP Express Router (SER) http://www.toyz.org/cgi-bin/sipwiki.cgi

Stephen [email protected]

Some more references

Look up the IETF RFCs on www.ietf.org, or google. Good primer: http://www.iptel.org/ser/doc/sip_intro/sip_introduction.h tml Wiki found on www.iptel.org. PINT is described in RFC 2458 and RFC 2848

Stephen [email protected]

SIP Fault Finding

SIP Workshop AARNet By Stephen Kingham [email protected]

Stephen Kingham

Outline and Objectives

Review different tools and particular call scenarios Ngrep for unix Ethereal for unix and MicroSoft SIP SAK (SIP Swiss Army Knife) SIPStone Server admin: Phpmyadmin to administer mysql data SER specific logs etc Monit and Big Brother DNS Review some faults User can not authenticate One way audio

Stephen [email protected]

NGREP for Unix

Perfect for running on SIP Server running on Unix platform. Allows you to watch traffic entering leaving the Server. Works because SIP protocol is in text and is user readable. Get from http://ngrep.sourceforge.net/ Hint: put in a carriage return at the end of each .. to make it easier to read. ALWAYS have it running, and keep it for a period of time, eg the last 3 days worth. That way faults reported after the fact can be investigated. Who in AARNet would like to write a script to do this? This script could get used around the world
Stephen [email protected]

Example from NGREP

interface: eth0 (192.94.63.0/255.255.255.128) match: sip U 192.94.63.73:3014 -> 192.94.63.28:5060 INVITE sip:[email protected] SIP/2.0.. Via: SIP/2.0/UDP 192.94.63.73:10038.. From: "61262112627" <sip:[email protected]>;tag=484b98c9-2db4 4512-95f9-aefc421085b8.. To: <sip:[email protected]>.. Call-ID: [email protected].. CSeq: 1 INVITE.. Contact: <sip:192.94.63.73:10038>.. User-Agent: Windows RTC/1.0..Content-Type: application/sdp.. Content-Length: 543.... v=0.. o=SRK-FIXED 0 0 IN IP4 192.94.63.73.. s=session.. c=IN IP4 192.94.63.73.. b=CT:1000..t=0 0.. m=audio 55586 RTP/AVP 97 111 112 6 0 8 4 5 3 101.. a=rtpmap:97 red/8000.. etc
Stephen [email protected]

Ethereal
Full protocol analyser for Unix, Mac OS, MicroSoft. Get ethereal from http://www.ethereal.com/ For MicroSoft Wondows install WinPcap first from the same location.

Stephen [email protected]

Ethereal

A Grab bag of tools

SIP SAK SIP Swiss Army Knife A traceroute like tool for SIP. Get from http://sipsak.berlios.de/ SIPStone Measuring SIP performance from Columbia University USA. See http://www.sipstone.com/ and/or http://www.cs.columbia.edu/IRT/cinema/sipstone MONIT and Big Brother Monit is a server watching utility which alerts administrators when a server dies. Big Brother is another server watching utility which alerts administrators when a server dies. Phpmyadmin to manage mysql tables Excellent for managing the SER Proxy Server data stored in the mysql data base. Get it from www.phpmyadmin.net.

Stephen [email protected]

DNS
SIP relies on DNS for routing (eg finding other SIP Servers). If something goes wrong with DNS then call setups can block for several seconds. Mitigate by: Cache DNS (eg nscd daemon in Linux) Have plenty of free children (threads) in the Proxy Server Process transactions statefully to absorb retransmissions without additional DNS lookups.

Stephen [email protected]

Hot topis
Slide 67

Stephen [email protected]