VoIP Basics. VoIP Basics, X/Stra, Oct 2,
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- Donald Leonard
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1 VoIP Basics 1
2 How did it all start? 2
3 How did it all start?
4 Encoding Media Pulse Code Modulation - a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a digital (usually binary) code. PCM has been widely used in: digital telephone systems compact disc red book format Computer systems (wav files). DVD or DVR Many Blu-ray Disc and HD-DVD movies Audio transmission within LANs Not used in real-time communication over the Internet due to high bandwidth consumption. emil.ivov@sip-communicator.org 4
5 Voice Codecs Codec Bit-Rate G711 64Kbs G ,3Kbs G729A 8Kbs ilbc 15.2 Kbs Speex variable 5
6 Transporting Media - TCP vs. UDP Using UDP No way to detect loss. Order of delivery does not necessarily reflect the order of sending Using TCP Loss detection Respects order Loss recovery inefficient for CoIP o Retransmission of lost segments increases jitter o Decreasing window size causes lower bandwidth emil.ivov@sip-communicator.org 6
7 UDP + RTP So, if both s**k, what do we do? Design a new transport protocol Design an application protocol that would compensate deficiencies of the transport protocol. A widespread solution Using an application protocol (RTP) over UDP emil.ivov@sip-communicator.org 7
8 What next?
9 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa emil.ivov@sip-communicator.org 9/47
10 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa emil.ivov@sip-communicator.org 10/47
11 The basics of IP telephony. network core (registrars, proxies, ) MEDIA Bob Address: B Port: Pb Alice Address: A Port: Pa emil.ivov@sip-communicator.org 11/47
12 Signaling Say why it is important to separate signalling from data. H.323 MGCP SIP XMPP/Jingle Other signaling protocols: IAX Skype, ICQ, Yahoo 12
13 MEGACO- H248-MGCP MGCP - Media Gateway Control Protocol Defines protocols for control of gateways that handle media flow conversion Example: transcoding analogous voice (PSTN) into digital signal IP. This approach is based on the notion of separating signaling from multimedia support. emil.ivov@sip-communicator.org 13
14 H248-MGCP (MEGACO) Media Gateway Controller PSTN IP Media Gateway PCM Flow 64 Kb/s RTP Flow 14
15 H.323 The ITU solution for video conferencing on data networks: IP, ATM, Strongly inspired by the RNIS H320 standards for conferencing. Multiple ITU PSTN low band protocols are employed by H.323 Q.931 Supplementary services coming from Q
16 H.323 ITU Recommendations Version : Video telephony system for LANs with no QoS Version 4 November 2000 Pros and Cons: Compatible with H320 (PSTN) High complexity, difficult to adapt to the Internet (Firewall, NATs, QoS) emil.ivov@sip-communicator.org 16
17 H.323 Entities Terminal Terminal Gatekeeper IP Network MCU Gateway RTC RNIS 17
18 Architectures and protocols RAS : Registration Admission Status Gatekeeper registration Q.931 : signaling call Allows opening an H.245 connection H.245 : control call Information exchange (codec, address, RTP and RTCP port numbers) Activates channels emil.ivov@sip-communicator.org 18
19 Ports Port Type Used for 389 static TCP ILS Registration (LDAP) 1300 static TCP H.235 Secure Signaling 1503 static TCP T static UDP Gatekeeper Discovery 1719 static UDP Gatekeeper RAS 1720 static TCP Q.931 Call Setup dynamic TCP dynamic UDP H245 Control Channel RTP/RTCP Audio/Video Streams emil.ivov@sip-communicator.org 19
20 A basic 2-party call UDP Q.931 port TCP 1720 H.245 port TCP > 1024 RTP G7xx RTCP RTP H26x RTCP RTP G7xx RTCP RTP H26x RTCP 20
21 The Zone T T T GW GW SCN MCU GK GW emil.ivov@sip-communicator.org 21
22 A Single Administrative Domain BE 22
23 Multiple Administrative Domains Clearing House Packet Network 23
24 VoIP Basics XMPP/Jingle 24
25 VoIP Basics The Session Initiation Protocol SIP 25
26 The Session Initiation Protocol (Some of the People Behind It) Henning Schulzrinne Department of Computer Science Columbia University, New York, USA Jonathan Rosenberg Cisco Systems 26
27 A Very Basic SIP Call Flow Alice Bob Transaction INVITE 100 Trying 180 Ringing 200 OK ACK MEDIA Dialog BYE OK CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 27
28 Example SIP Request INVITE SIP/2.0 Via: SIP/2.0/UDP ;branch=1 From: To: Contact: Call-ID: Cseq: INVITE v=0 o=user IP s=. t=0 0 c=in IP m=audio RTP/AVP 0 a=rtpmap:0 PCMU/
29 Example SIP Request Request line INVITE SIP/2.0 Headers Via: SIP/2.0/UDP ;branch=1 From: To: Contact: Call-ID: Cseq: INVITE Empty line Body v=0 o=user IP s=. t=0 0 c=in IP m=audio RTP/AVP 0 a=rtpmap:0 PCMU/
30 Example SIP Response SIP/ Not Found Via: SIP/2.0/UDP ; branch=1 From: To: Call-ID: Cseq: INVITE 30
31 Example SIP Response Response line SIP/ Not Found Headers Via: SIP/2.0/UDP ; branch=1 From: To: Call-ID: Cseq: INVITE Empty line 31
32 Intra Domain SIP Signaling 4 thomas@u-strasbg.fr 5 SIP Soft Client 3 7 emcho@u-strasbg.fr 6 2 u-strasbg.fr Proxy Server 1 SIP Phone u-strasbg.fr Registrar and Location Service 1. Call Thomas - INVITE 2. Query Where is Thomas@u-strasbg.fr? (non-sip) 3. Response (non-sip) 4. Proxied Call - INVITE 5. Response - OK 6. Response - OK 7. Multimedia Chanel Establised RTP Streams CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 32
33 Registrars and Registrations LittleGuy Registrar REGISTER From: LittleGuy Contact: 200 OK Contact: CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 33
34 Registrars and Registrations LittleGuy Registrar REGISTER From: LittleGuy Contact: 401 Unauthorized Contact: WWW-Authenticate: <Authentication Challenge> REGISTER From: LittleGuy Contact: Authorization: <Authentication Response> 200 OK Contact: CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 34
35 An example REGISTER request REGISTER sip:b.com SIP/2.0 Via: SIP/2.0/UDP From: To: Contact: Expires: 3600 Call-ID: CSeq: 1 REGISTER CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 35
36 An example REGISTER request Request-URI registration domain Who s registering AOR Contact REGISTER sip:b.com SIP/2.0 Via: SIP/2.0/UDP From: sip:barbara@b.com;tag= To: sip:barbara@b.com Contact: <sip:barbara@ > Duration in seconds Expires: 3600 Empty line Call-ID: @ CSeq: 1 REGISTER CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 36
37 An example REGISTER response SIP/ Ok Via: SIP/2.0/UDP From: To: es=3600 Contact:<sip: >;expires=345 Contact:<sip: >;expires=1000 Call-ID: CSeq: REGISTER CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 37
38 An example REGISTER response SIP/ Ok Via: SIP/2.0/UDP Who s regisering AOR List of all Contact headers for know AORs From: sip:barbara@b.com;tag= To: sip:barbara@b.com;tag=jjf223 Contact:<sip:barbara@ >;expir es=3600 Contact:<sip: >;expires=345 Contact:<sip: >;expires=1000 Call-ID: @ CSeq: REGISTER Empty line CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 38
39 REGISTER: refresh, cancel, query It is up to the user agent to refresh registrations of Contact addresses. In order to do so, a UA has to resend its innitial REGISTER request. In order to cancel a Contact registration, a user agent has to set its Expires time to zero To: sip:barbara@b.com Contact: <sip:barbara@ > Expires: 0 In order to cancel all contact address of records, a UA could use an asterisk (*) To: sip:barbara@b.com Contact: * Expires: 0 Omitting the Contact header would not modify any AOR and the corresponding response would contain all existin AORs. CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 39
40 Inter Domain SIP Signaling Transaction INVITE 1 INVITE 2 TRYING 3 INVITE 4 TRYING 5 RINGING 6 RINGING 8 RINGING 7 OK 9 OK 10 OK 11 ACK MEDIA Dialog BYE OK CoIP with SIP. Rennes, November 2007 Emil Ivov, SIP Communicator 40
41 Reality Check Hellooo did u forget about NATs? emil.ivov@sip-communicator.org 41/47
42 Standard NAT usage NAT 42
43 Less standard NAT usage: End to end services NAT? NAT 43
44 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa emil.ivov@sip-communicator.org 44/47
45 The basics of IP telephony. network core (registrars, proxies, ) Bob Address: B Port: Pb Alice Address: A Port: Pa emil.ivov@sip-communicator.org 45/47
46 The basics of IP telephony. network core (registrars, proxies, ) MEDIA Bob Address: B Port: Pb Alice Address: A Port: Pa emil.ivov@sip-communicator.org 46/47
47 And then NATs were born Call: To: B Media: Ap Call: To: B Media: Ap ERROR Alice Private Address: Ap NAT/Firewall Address: F Bob Address: B emil.ivov@sip-communicator.org 47/47
48 How do NATs work Internal host:port NAT port : MSG: Dst: : 80 Src: : 2368 MSG: Dst: : 80 Src: : 8632 Alice NAT Internal Address: NAT Public Address: Server Address: emil.ivov@sip-communicator.org 48
49 How do NATs work Internal host:port NAT port : MSG: Dst: : 2368 Src: : 80 MSG: Dst: : 8632 Src: : 80 Alice NAT Internal Address: NAT Public Address: Server Address: emil.ivov@sip-communicator.org 49
50 How do NATs work Internal host:port NAT port : Endpoint-Independent Mapping Endpoint-Independent Filtering Bob Address: MSG: Dst: : 2368 Src: : 9595 Alice NAT Internal Address: NAT Public Address: Server Address: emil.ivov@sip-communicator.org 50
51 Basic Firewall and NAT Traversal STUN What are my address and port? Address: F Port: Pf STUN Server Alice Address: Ap Port: Pa NAT Address: F Bob Address: B Call: To: B Media: F:Pf STUN Server Alice Address: Ap Port: Pa Answer: To: A Media: B emil.ivov@sip-communicator.org Bob Address: B 51
52 How do NATs work Address (and port) dependent filtering Internal host:port NAT port Active connections host:port : (: 80) MSG: Dst: : 80 Src: : 2368 MSG: Dst: : 80 Src: : 8632 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 52
53 How do NATs work Address (and port) dependent filtering Internal host:port NAT port Active connections host:port : (: 80) MSG: Dst: : 2368 Src: : 80 MSG: Dst: : 8632 Src: : 80 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 53
54 How do NATs work Address (and port) dependent filtering Internal host:port NAT port Active connections host:port : (: 80) Endpoint-Independent Mapping Endpoint-Dependent Filtering Bob Address: STOP Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 54
55 How do NATs work Address (and port) dependent filtering Internal host:port NAT port Active connections host:port : (: ) (: 80) Bob Address: MSG: Dst: : 80 Src: : 2368 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 55
56 How do NATs work Address (and port) dependent filtering Internal host:port NAT port Active connections host:port : (: ) (: 80) Endpoint-Independent Mapping Endpoint-Dependent Filtering Bob Address: MSG: Dst: : 2368 Src: : 80 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 56
57 How do NATs work Endpoint dependent mapping Internal host:port NAT port Active connections host:port : (: 80) MSG: Dst: : 80 Src: : 2368 MSG: Dst: : 80 Src: : 8632 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 57
58 How do NATs work Endpoint dependent mapping Internal host:port NAT port Active connections host:port : (: 80) MSG: Dst: : 2368 Src: : 80 MSG: Dst: : 8632 Src: : 80 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 58
59 How do NATs work Endpoint dependent mapping Internal host:port NAT port Active connections host:port : (: 80) : (: 80) Bob Address: MSG: Dst: : 80 Src: : 2368 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 59
60 How do NATs work Endpoint dependent mapping Internal host:port NAT port Active connections host:port : (: 80) : (: 80) Endpoint-Dependent Mapping Endpoint-Dependent Filtering Bob Address: MSG: Dst: : 2368 Src: : 80 Alice NAT Internal Address: NAT Public Address: STUN Server Address: emil.ivov@sip-communicator.org 60
61 Relaying Media Symmetric NAT/Firewall F1:P1 STUN Relay Server Address: T Port: Pt Alice Address: Ap Port: Pa Call: To: B Media: T:Pt Symmetric NAT/Firewall F1:p2 Bob Address: B emil.ivov@sip-communicator.org 61
62 Relaying Media Symmetric NAT/Firewall F1:P1 STUN Relay Server Address: T Port: Pt Bob Address: B Alice Address: Ap Port: Pa Symmetric NAT/Firewall F1:p2 emil.ivov@sip-communicator.org 62
63 Relaying Media (The SIP Way) NAT/Firewall SIP Server Address: T Port: Pt Alice Address: Ap Port: Pa Bob Address: B emil.ivov@sip-communicator.org 63/47
64 Relaying Media (The SIP Way) NAT/Firewall SIP Server Address: T Port: Pt Alice Address: Ap Port: Pa Bob Address: B emil.ivov@sip-communicator.org 64/47
65 Relaying Media SIP clients behind a symmetric NAT/firewall non-scalable expensive complex symmetric firewall Relay Server SIP clients behind a symmetric NAT/ firewall symmetric NAT/firewall emil.ivov@sip-communicator.org 65/47
66 Could we please have IPv6 now? ok, it s probably high time we moved to IPv6 emil.ivov@sip-communicator.org 66/47
67 Could we please have IPv6 now? this should simplify VoIP shouldn t it? emil.ivov@sip-communicator.org 67/47
68 VoIP and IPv6 demo version network core (registrars, proxies, ) Bob 2001:660::1 Alice 2001:660::2 emil.ivov@sip-communicator.org 68/47
69 VoIP and IPv6 demo version network core (registrars, proxies, ) Bob 2001:660::1 Alice 2001:660::2 emil.ivov@sip-communicator.org 69/47
70 VoIP and IPv6 demo version network core (registrars, proxies, ) MEDIA Bob 2001:660::1 Alice 2001:660::2 emil.ivov@sip-communicator.org 70/47
71 Reality check! Reality check! 71/47
72 Reality check! VPN Priv: Pub: Stun Relay Server Alice 2001:660:: NAT SIP network Bob Alice s list of addresses: 2001:660:: /47
73 How to avoid relaying? Interactive Connectivity Establishment (ICE) An IETF draft brought to you by Cisco s Jonathan Rosenberg emil.ivov@sip-communicator.org 73/47
74 Address management with ICE VPN Priv: Pub: Stun Relay Server Alice NAT SIP network Bob Alice s list of addresses: 2001:660:: Please try me on any of the following: 2001:660:: /47
75 Address management with ICE ERROR ERROR VPN Priv: Pub: Stun Relay Server Alice NAT SIP network Bob ERROR Alice s list of addresses: 2001:660:: ERROR ERROR 2001:660::2 75/47
76 IP Telephony Deployment 76/47
77 Asteriskv6.org (*) Reliability support for failover A lot of available resources Scalability Simple Configurations ~ 30 calls Soekris board, no transcoding ~ 60 calls with transcode Pentium4 2.4 = 80% busy ~ Up to 250 lines through PRI Digium cards (zaptel limitation) Clustering Load sharing Combining with SER Codecs: ADPCM, G.711 (A-Law & µ-law), G. 722, G (pass through), G. 726, G.729, GSM, ilbc, Linear, LPC-10, Speex Other features: conference bridging voice mail (including delivery) echo and MP3 plugins call parking, queueing, recording, retrieving, snooping caller id call blocking ENUM fax transmit and receive music on hold, and transfer. emil.ivov@sip-communicator.org 77
78 SIP Express Router (SER) Light weight Very fast Routing wise, more sophisticated than (*) (routing can be based on packet contents) Packet rewriting. Modules are available for: accounting, authentication, interaction with RADIUS, ENUM, SIMPLE, NAT Support, SMS gateway, web interface, stateless replies, presence agent, MySQL interaction, Jabber interaction Support for IPv6 78
79 SIP Express Media Server (SEMS) Especially well suited for work with SER Supports plugins Plugins shipped with SEMS: Voic record messages and mail them. ISDN Gateway: support calls from and to the PSTN. Conferencing: connect people within a conference room. Announcement: plays an announcement. Echo: test module echoing your voice. Codecs: G711.u, G711.a, GSM emil.ivov@sip-communicator.org 79
80 Open SER Small footprint Plug&Play module interface - ability to add new extensions, without touching the core support for UDP/TCP/TLS transport layers IPv4 and IPv6 Flexibility of routing configuration authentication, authorization and accounting (AAA) CPL - Call Processing Language (RFC3880) NAT traversal support for SIP and RTP traffic ENUM support Extension interfaces for PERL and Javaload balancing with failover, support for replication Interconnection with PSTN, XMPP, SMS multiple database backends - MySQL, PostgreSQL, over 70 extra modules in the OpenSER repository emil.ivov@sip-communicator.org 80
81 Open SER is no longer OpenSIPS.org Kamailio.org 81
82 Media relaying with RTP Proxy High performance RTP stream proxy-ing Works with SER OpenSER through their nat helper module Supports features such as: Remote control mode IPv4 IPv6 IPv4 to IPv6 relaying 82/47
83 Clients the other side 83/47
84 SIP Communicator Overview Audio/Video Calls with SIP 84/47
85 SIP Communicator Overview Audio/Video Calls with SIP 85/47
86 SIP Communicator Overview Instant Messaging 86/47
87 Other? X-Lite KPhone Linphone Hardware clients: 87/47
88 A Sample Deployment Strasbourg Sofia ULP PSTN Network Sofia Public STN Digium Wildcard IPv4 Linksys PAP Linksys SPA3000 VoIP Phones IPv4 IPv6 PSTN access, transcoding, conferencing RTPPROXY IPv6 <-> IPv4 for RTP IAX IPV6 Registrar VoIP Phone voipgw.u-strasbg.fr SIP Communicator SIP Communicator 88
89 VoIP Basics 89
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