1. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SIP: Session Initiation Protocol

2. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Signaling Protocol Review H323: ITU peer:peer protocol. ISDN (Q.931) signaling stuffed into packets. Can be TCP or UDP. –H225: Q931 for call control, RAS to resolve endpoints (Gateway  Gatekeeper signaling) –H245: Multimedia Control Protocol (including fast start) –Uses RTP for media MGCP (MEGACO): IETF Master:slave,very granular messages, typically used by softswitches to control endpoints, media gateways, etc. Also uses RTP for media. SIP: IETF peer-peer protocol. Plain text, messages look like http. –SDP (Session Description Protocol) used to negotiate RTP. –DNS (SRV records) used to resolve URIs –Typically UDP (except for secure SIP, which is usually SIP running over TLS/TCP

3. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SIP: Related Protocols Secure SIP: –SIP:TLS (transport layer security): UDP –IP-SEC (used typically only between servers) –S/MIME (interesting solution for end-end encryption of portions of the SIP message, such as the encryption keys for SRTP). Media: –SDP: negotiates codecs, ip_address:port, etc. Carried in the SIP message. –RTP: time stamps & sequence # (jitter, lost packets), packet type (what type of codec). –SRTP: encrypted RTP, keys not negotiated in the media –ZRTP: (encrypted RTP (Zimmerman’s verison), keys negotiated in the media channel. –RTCP: quality and control channel for RTP (port=RTP+1) –H.264 (and 263): Video compression DNS: SRV records, ENUM STUN & ICE: NAT traversal.

4. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SIP Network Elements User Agent: endpoints that originate or terminate SIP calls => IP Phones, Softclients, etc. SIP Servers: –Proxy/Redirect: routing of SIP messages –Location: endpoints register here, binds IP address to URIs & telephone numbers –Back:Back User Agents: terminate and re-launch SIP message (often used by IP PBXs, SBCs, feature servers, etc.) –Feature Servers: SIP interface, used to provide functionality (calling services, voicemail, conferencing, credit card, etc.) Media Gateways: –PSTN  VoIP interworking: –Media: RTP to B channel –Signaling: SIP or MGCP to Q.931(PRI) to ISUP(SS7),

5. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SIP Methods (from voip-info.org) SIP methods defined in the SIP RFC (3261) SIP method invite : Invite another UA to a sessionSIP method invite SIP method invite re-invite: Change UA within an existing sessionSIP method invite re-invite SIP method register: Register a location with a SIP Registrar serverSIP method registerSIP Registrar server SIP method ack: Used to facilitate reliable message exchange for INVITEsSIP method ack SIP method cancel: Cancel an inviteSIP method cancel SIP method bye: Hangup a sessionSIP method bye SIP method options: Query/response of UA capabilities (codecs, etc.)SIP method options SIP method extensions from other RFCs SIP method info: out of band signaling info, like DTMF RFC 2976SIP method infoRFC 2976 SIP method notify: report events, RFC 2848 PINTSIP method notifyRFC 2848PINT SIP method subscribe: subscribe to events, RFC 2848 PINTSIP method subscribeRFC 2848PINT SIP method unsubscribe: duh RFC 2848 PINTSIP method unsubscribeRFC 2848PINT SIP method update: Extension in RFC 3311SIP method updateRFC 3311 SIP method message: Extension in RFC 3428SIP method message SIP method refer: 3 rd party call control, transfer, RFC 3515SIP method refer SIP method prack: provisional ACK, intermediate response to ACK, RFC 3262SIP method prack SIP method PUBLISH: think this is like a NOTIFY RFC 3903RFC 3903 SIP SIMPLE: Instant Messaging implemented using SIP (competing protocol is xmpp, also proprietary protocols like AOL.)

6. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SIP Header Fields INVITE sip:7816466711@mit.edu SIP/2.0 Via: SIP/2.0/UDP 209.150.62.132:22164;rport;branch=z9hG4bK4CD60C37023D41A08529035ECA54ADAD From: Hillary Clinton ;tag=1217297847 To: Contact: Call-ID: D98ADF5F-AF84-429E-A1A0-4D8CC34D66FA@192.168.0.4 CSeq: 23513 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105x Content-Length: 318 Method = INVITE, uses SIP version 2.0 VIA = How the message is routed (when using record route, you get full list of servers the call is routed through). Contact: SIP URL of the guy sending the message (usually = FROM) Call-ID: each session (call) has a unique call-ID. Used to provide state within a session, also for accounting. CSeg: +1 for each request within a session User Agent: user agent name & version Content-Length: number of bytes in the message

7. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SDP: Session Description Protocol v: version o: origination info, s: session name t: times c: Connection info m: media session <list of fmtp types, e.g., media types, that the client can support a: attrbutes as pairs k: encryption key (can be here, or can be carried in the SIP message) v=0 o=- 3515669133 3515669133 IN IP4 18.8.202.4 s=SIP Call c=IN IP4 18.8.202.4 t=0 0 m=audio 23400 RTP/AVP 0 101 c=IN IP4 18.8.202.4 a=fmtp:101 0-16 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000

8. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Simple Call Flows – Hello World

9. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Hello World – Ladder Diagram

10. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Example: Call goes to voicemail – with MWI Extension 3004 calls extension 3002 through an Asterisk IP-PBX (acting as a B:B UA) Extension 3002 is busy. Asterisk moves the call to itself – to the voicemail server on the box Voicemail is left (RTP between ext 3004 and the asterisk server). Asterisk sends a SIP NOTIFY with event type = MWI (message waiting indicator) to ext 3002

11. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.101:5060 -> 192.168.0.110:5060 INVITE sip:3002@192.168.0.110;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.0.101:5060. From: ;tag=705048313. To:. Call-ID: 4107902737@192.168.0.101. CSeq: 2 INVITE. Contact:. User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A). Proxy-Authorization: Digest username="3004",realm="asterisk",nonce="1fdb7e7b",uri="sip:3002@192.168.0.110",response="a1bb3a7501b0982650a93 bffacca0568". Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER. Content-Length: 247. Content-Type: application/sdp.. v=0. o=3004 1882 1882 IN IP4 192.168.0.101. s=ATA186 Call. c=IN IP4 192.168.0.101. t=0 0. m=audio 10002 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.

12. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.110:5060 -> 192.168.0.101:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.0.101:5060. From: ;tag=705048313. To: ;tag=as6ce511fd. Call-ID: 4107902737@192.168.0.101. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact:. Content-Length: 0.

13. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.110:5060 -> 192.168.0.105:5060 INVITE sip:3002@192.168.0.105 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK06dae12e. From: "3004" ;tag=as39b4da84. To:. Contact:. Call-ID: 637e4707796938ad07d4e39a7e6ef404@192.168.0.110. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Sun, 14 Nov 2004 19:19:49 GMT. Alert-info: Bellcore-dr1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Content-Type: application/sdp. Content-Length: 238.. v=0. o=root 28925 28925 IN IP4 192.168.0.110. s=session. c=IN IP4 192.168.0.110. t=0 0. m=audio 22062 RTP/AVP 0 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16.

14. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.105:5060 -> 192.168.0.110:5060 SIP/2.0 486 Busy here. Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK06dae12e. From: "3004" ;tag=as39b4da84. To: ;tag=000628f0f9f2000d7a71ad55-38f1a626. Call-ID: 637e4707796938ad07d4e39a7e6ef404@192.168.0.110. Date: Sun, 14 Nov 2004 19:19:49 GMT. CSeq: 102 INVITE. Server: CSCO/4. Contact:. Content-Length: 0. U 192.168.0.110:5060 -> 192.168.0.105:5060 ACK sip:3005@192.168.0.105:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK06dae12e. From: "3004" ;tag=as39b4da84. To: ;tag=000628f0f9f2000d7a71ad55-38f1a626. Contact:. Call-ID: 637e4707796938ad07d4e39a7e6ef404@192.168.0.110. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0.

15. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.110:5060 -> 192.168.0.101:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.101:5060. From: ;tag=705048313. To: ;tag=as6ce511fd. Call-ID: 4107902737@192.168.0.101. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact:. Content-Type: application/sdp. Content-Length: 237.. v=0. o=root 28925 28925 IN IP4 192.168.0.110. s=session. c=IN IP4 192.168.0.110. t=0 0. m=audio 5812 RTP/AVP 0 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16.

16. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.101:5060 -> 192.168.0.110:5060 ACK sip:3002@192.168.0.110 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.101:5060. From: ;tag=705048313. To: ;tag=as6ce511fd. Call-ID: 4107902737@192.168.0.101. CSeq: 2 ACK. User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A). Proxy-Authorization: Digest username="3004",realm="asterisk",nonce="1fdb7e7b",uri="sip:3002@192.168.0.110",response="a1bb3a7501b0982650a93bffacca0568". Content-Length: 0..101:10002.110:5812

17. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. U 192.168.0.110:5060 -> 192.168.0.105:5060 NOTIFY sip:3002@192.168.0.105 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK361f812f. From: "asterisk" ;tag=as39e043e0. To:. Contact:. Call-ID: 3a86cecd339da70710f761f11e0515d4@192.168.0.110. CSeq: 102 NOTIFY. User-Agent: Asterisk PBX. Event: message-summary. Content-Type: application/simple-message-summary. Content-Length: 37.. Messages-Waiting: yes Voicemail: 2/0 # U 192.168.0.105:50268 -> 192.168.0.110:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK361f812f. From: "asterisk" ;tag=as39e043e0. To:. Call-ID: 3a86cecd339da70710f761f11e0515d4@192.168.0.110. Date: Sun, 14 Nov 2004 19:20:04 GMT. CSeq: 102 NOTIFY. Content-Length: 0.

18. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. So, you can’t tell but Asterisk server is sending this call to voicemail. Media goes to the asterisk server And now the asterisk server sends ext 105 a MWI notification.

19. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SIP: How to Authenticate the sender of a SIP message? Digest Authentication – from the http world Assumption: shared secret (sip password) between endpoints exists (generally passed to the endpoint during the provisioning process). Initial method rejected – for example, recipient replies with –Proxy Authentication Required –WWW-Authenticate: Digest realm="mit.edu", nonce="47a23e53632ff1c28b41f12591b970586d6 da3dc“ Second try uses a hash of the nonce, the passwd, realm, & other fields to form a response which validates the identity of the sender –Authorization: Digest username="6173249175", realm="mit.edu",nonce="47a23e53632ff1c28b41f 12591b970586d6da3dc",response="fbab4f8c828c a550da0956d648d8d09a",uri="sip:mit.edu" Invite 407 Proxy Auth Req, with nonce ACK Invite with response ok

20. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Digest Authentication: SIP Decision of which sip messages to challenge is up to the system (beauty in the eye of the system architect). –INVITES –REGISTERS –BYES –REFERS –ACKS Digest Authentication provides authentication and immunity from replay attacks. It does not provide any privacy or guarantee message integrity. –SIP messages still in plain text, can be read. –Man in the middle could change some of the fields in a SIP message Other forms of encryption (TLS, IP-SEC, S/MINE, etc.) provide security of the SIP message streams. Typical Deployments Today –Often the network proxy server requires authentication of the endpoints for Registration and Invites. –Often the endpoints (phones, clients) DO NOT require authentication of the servers. (Very often, an ACL on the endpoints is used to prevent the endpoint from replying to “bad guys.”

21. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Generating the digest response HA1 = MD5{username +”:”+realm+”:”+password} –Username = 6173249175, realm = mid.edu, passwd is the secret not passed in this transaction HA2 = MD5{“sip:”+ uri from the Request-URI sip header} –uri=sip:mit.edu Response = MD5{HA1 +”:”+ nonce +”:” +nonceCount+”:”+clientNonce+”:”+qop+”:”+HA2} –Nonce = unique string generated by the challenger –Other stuff – not so important MD5 = “one-way” hash, designed to be too hard (for everyone except the folks with giant super computers?) to guess the output based upon the input.

22. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Replay Attacks Basically, generate a nonce each time you want to challenge the message = or a “bad guy” could simply respond with the same old stuff. Believe it or not, I’ve seen systems that were subject to replay attacks.

23. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Summary username="6173249175“ realm="mit.edu" nonce="47a23e53632ff1c28b41f12591b97 0586d6da3dc" uri="sip:mit.edu" sip passwd = response="fbab4f8c828ca550da0956d648 d8d09a“ Response = hard function (BLUE, RED)

24. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Another SIP Example: Shared Call Appearance (Bridged Line Appearance)

25. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. SUBSCRIBE & NOTIFY: When Mr. Burn’s picks up the phone His line key turns red, to show that he’s actively on the line Smither’s line blinks red, to show him that Mr. Burns is on the line 6172531234

26. Copyright 2005 – 2008 © by Elliot Eichen. All rights reserved. Subscribe/Notify Call Flows