sip

37
Introduction to SIP deployment TNC2004 VoIP workshop Rodos, 11 June 2004 Erik Dobbelsteijn [email protected]

Upload: sandra4211

Post on 10-Jun-2015

750 views

Category:

Documents


0 download

TRANSCRIPT

Page 1: SIP

Introduction to SIP deployment

TNC2004 VoIP workshop

Rodos, 11 June 2004

Erik [email protected]

Page 2: SIP

2

Agenda

• What can you do with SIP• What is SIP• SIP and H.323• VoIP deployment schemes• Basic deployment• Extended deployment• Client configuration• Session example• Issues• Looking forward• Referals

Page 3: SIP

3

Your Deployment Bible:

http://www.terena.nl/tech/IPtel

Page 4: SIP

4

Group communication on internet

Instant Messaging, chat

Voice over IP

Video conf. Desktop- sharing,Application- sharing

E-mail, forum,Listserv,News

Voice-mail

Video-mail,Streaming

.ppt, .doc attachment

text audio video

synchronous

a-synchronous

data

Page 5: SIP

5

What can you do with SIP now

• Set up a real-time communication session in an e-mail kind of fasion

• Voice, or Video & Voice• Call for free from IP to IP• Cheap deployment• Set up gateway to POTS for calls to regular

phones• Integrate with messaging

Page 6: SIP

6

What is SIP

• Deals with signaling for setting up a real-time communication session

• Messages are HTTP like (plain text, not binary)• Registration of User Agents (‘telephones’) at

SIP Registrar• Call handling by SIP proxy• Audio and/or videostreams follow straight

path between User Agents

Page 7: SIP

7

SIP and H.323

• Religious war: think your requirements through, first

• SIP++: Use it like e-mail, forget the number!• SIP++: Find each other through DNS• SIP++: Multi-client support• SIP++: Easy integration with messaging

Page 8: SIP

8

VoIP deployment schemes

• PABX replacement

• Campus population (SIP.edu)

• Inter-office switching (trunking)

• Public internet service (Pulver Free World Dialup service: http://fwd.pulver.com)

Page 9: SIP

9

Basic deployment

• Calls between IP phones and/or Soft phones

– Each User Agent (UA) registers with Registrar

– Proxy forwards call requests to correct UA

Page 10: SIP

10

Basic Setup

LAN

SIP

proxy

SIP

registrar

Software UA

Hardware UA

PCPC

Hardware UA

Page 11: SIP

11

Step 1: registering client

LAN

SIP

proxy

SIP

registrar

Software UA

Hardware UA

PCPC

Hardware UA

REGISTER sip:192.87.116.77 SIP/2.0Via: SIP/2.0/UDP 192.168.0.134:5060To: Erik Dobbelsteijn<sip:[email protected]>From: Erik Dobbelsteijn<sip:[email protected]>Call-ID: [email protected]: 24297 REGISTERMax-Forwards: 70Expires: 3600Contact: <sip:[email protected]:5060>

SIP/2.0 200 OK

Page 12: SIP

12

Step 2: call setup

LAN

SIP

proxy

SIP

registrar

Software UA

Hardware UA

PCPC

Hardware UA

INVITE [email protected]

Multiple

UA’s

Can answer

200 OK

Page 13: SIP

13

Step 3: voice and video streams

LAN

SIP

proxy

SIP

registrar

Software UA

Hardware UA

PCPC

Hardware UA

Audio

(+video)

stream

Page 14: SIP

14

Basic deployment issues (to do list)

• Hardphones and/or Soft phones?• Which SIP Proxy Manufacturer?• Dialplan• Authentication• NAT/Firewall • Redundancy• Logging• End-user support

Page 15: SIP

15

Soft phones & hard phones

+/+ flexible+/+ cheap+/+ messaging-/- audio setup-/- always on

+/+ audio setup+/+ always on-/- more expensive-/- messaging

Page 16: SIP

16

Manufacturer

• Free (Open Source):– SER (www.iptel.org/ser)– Asterisk (www.asterisk.org)

• Commercial:– See www.sipcenter.com or www.pulver.com– Microsoft Live Communications Server– Wave3 Sessions– Radvision ViaIP

Page 17: SIP

17

Dialplan

• Keep it simple: identify users by e-mailadres

• To support easy in/outside dialing: make number aliases: ‘real’ telephone number, consisting of:

– short number– Prefix

(+31302305)309 = [email protected]

• Process numbers: add/strip prefix

Page 18: SIP

18

Authentication

• basic vs Digest (MD5): use Digest• TLS/Kerberos: interop problems

Re-use existing backend:• RADIUS• LDAP

Page 19: SIP

19

NAT/Firewall

Main problem: dynamic UDP ports for media

Campus deployment:• Probably no NAT necessary/implemented • Firewall: keep all SIP components behind firewall• Firewall: if SIP server needs to be protected from

campus infrastructure: enable port 5060. Audio streams go between UAs directly, so no firewall issue

• NAT traversal: see Interactive Connectivity Establishment (ICE)

Page 20: SIP

20

Redundancy

• Double Registrar• Double Proxy• Double RADIUS• Double LDAP• (host userdata database and webserver

separately)

Page 21: SIP

21

Logging

• Text based logging: REGISTER, INVITE, OK etc

• Accounting to RADIUS

• Accounting to PSTN: get it out of PABX

Page 22: SIP

22

Extended deployment

• DNS service records: resolve SIP UA locations• Voicemail to E-mail• End-user self provisioning• Integration with Messaging (SIMPLE/JABBER)• Click to Dial• Calls to POTS (a.k.a. PSTN)• MultiConference Unit (for multi party calls)• ENUM: resolve POTS numbers to IP domain• Interfacing with H.323• Multi-domain hosting

Page 23: SIP

23

DNS Service records

• Find users’ Proxy like mailservers do: by DNS Service Record

_sip._udp.surfnet.nl server selection 0 0 5060 sip.showcase.surfnet.nl

If a SIP request for *@surfnet.nl arrives over UDP at port 5060…

…ask this server

Page 24: SIP

24

Web based self provisioning (SER)

• Configure a mailserver and local message handling (restricted sendmail)

• Install Apache and PHP• Install SER web

Page 25: SIP

25

Web based self provisioning (SER)

Page 26: SIP

26

Integration with Messaging (SER)

• SIMPLE is integrated and enabled by default: test it with Microsoft Windows Messenger 4.7

• SER can act as a SIMPLE to JABBER gateway

Page 27: SIP

27

POTS gateway

LAN

Software UA

SIP

proxy

Hardware UA

SIP

registrar

Gateway

POTS

SIP

User Agents

PCPC

Hardware UA

PABX

Page 28: SIP

28

POTS gateway• Provides call handling and media conversion

towards PSTN• If number is not a short number/starts with a

0: send call request to gateway• PC or router with ISDN2/ISDN30 interfaceCisco:dial-peer voice 112 voip destination-pattern 31302305... session target sip-server no vadsip-ua nat symmetric role passive nat symmetric check-media-src retry invite 4 retry response 3 retry bye 2 retry cancel 2 sip-server dns:surfnet.nl

Page 29: SIP

29

SIP-H.323 gatewaying

• Not quite stable yet• Easiest way: use a mixed-protocol gateway or

MCU.

Page 30: SIP

30

MultiConference Unit (for multi party calls)

• Mix audio and video signals• MCU is registered at the Proxy• Mixed H.323/SIP available (RADVision, FVC)

Multi Conference Call

Page 31: SIP

31

ENUM

• Resolve PSTN numbers (E.164) to IP• +31 30 2305109 will go first to SIP phone,

then to H.323 phone and e-mail otherwise

$ORIGIN 9.0.1.5.0.3.2.0.3.1.3.e164.arpaIN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:[email protected]!" .IN NAPTR 10 101 "u" "E2U+h323" "!^.*$!h323:[email protected]!" .IN NAPTR 10 102 "u" "E2U+msg:mailto" "!^.*$!mailto:[email protected]!" .

Page 32: SIP

32

Multi-Domain hosting (SER)

if (uri=~"surfnet.nl") {route(1);break;

} else if (uri=~"chat.surfnet.nl") {route(2);break;

} else if (uri=~"sip.showcase.surfnet.nl") {route(3);break;

};

Page 33: SIP

33

Client configuration

• Accountname, sign-in name, password

Page 34: SIP

34

Session example

• Start from the menu, or the Buddy List:

Page 35: SIP

35

Issues

• Interoperability (single vendor can be good choice

• MS Windows Messenger 4.7/ 5 /MSN mixup• Presence/IM not widely supported yet• PC based configuration: instable audio setup

(mute button, headphone not plugged in etc)

Page 36: SIP

36

The (near) future:

• Context aware and Location Based communication (see http://pic.internet2.edu)

• Extended reachability via database (H.350)• Added security and anti-spam• IPv6

Page 37: SIP

37

For rainy Sunday afternoons…

TERENA VoIP cookbook: http://www.terena.nl/tech/voip

SIP.edu cookbook: http://voip.internet2.edu/SIP.edu

SURFnet VoIP pages (Dutch)http://www.surfnet.nl/innovatie/surfworks/voip

http://www.sipcenter.comhttp://fwd.pulver.com