sip
TRANSCRIPT
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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
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Your Deployment Bible:
http://www.terena.nl/tech/IPtel
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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
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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
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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
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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
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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)
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Basic deployment
• Calls between IP phones and/or Soft phones
– Each User Agent (UA) registers with Registrar
– Proxy forwards call requests to correct UA
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Basic Setup
LAN
SIP
proxy
SIP
registrar
Software UA
Hardware UA
PCPC
Hardware UA
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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
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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
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Step 3: voice and video streams
LAN
SIP
proxy
SIP
registrar
Software UA
Hardware UA
PCPC
Hardware UA
Audio
(+video)
stream
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Basic deployment issues (to do list)
• Hardphones and/or Soft phones?• Which SIP Proxy Manufacturer?• Dialplan• Authentication• NAT/Firewall • Redundancy• Logging• End-user support
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Soft phones & hard phones
+/+ flexible+/+ cheap+/+ messaging-/- audio setup-/- always on
+/+ audio setup+/+ always on-/- more expensive-/- messaging
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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
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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
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Authentication
• basic vs Digest (MD5): use Digest• TLS/Kerberos: interop problems
Re-use existing backend:• RADIUS• LDAP
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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)
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Redundancy
• Double Registrar• Double Proxy• Double RADIUS• Double LDAP• (host userdata database and webserver
separately)
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Logging
• Text based logging: REGISTER, INVITE, OK etc
• Accounting to RADIUS
• Accounting to PSTN: get it out of PABX
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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
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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
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Web based self provisioning (SER)
• Configure a mailserver and local message handling (restricted sendmail)
• Install Apache and PHP• Install SER web
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Web based self provisioning (SER)
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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
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POTS gateway
LAN
Software UA
SIP
proxy
Hardware UA
SIP
registrar
Gateway
POTS
SIP
User Agents
PCPC
Hardware UA
PABX
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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
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SIP-H.323 gatewaying
• Not quite stable yet• Easiest way: use a mixed-protocol gateway or
MCU.
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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
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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]!" .
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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;
};
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Client configuration
• Accountname, sign-in name, password
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Session example
• Start from the menu, or the Buddy List:
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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)
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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
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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