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IT-Symposium 2005 7. April 2005 VoIP mit Asterisk PBX Voice over - PDF document

IT-Symposium 2005 7. April 2005 VoIP mit Asterisk PBX Voice over IP Telephonie mit Asterisk Nebenstellenanlagen John R. Covert DECUS-Mnchen Symposium 2005 3E06 VoIP Technologiesituation Netzprotokolle Asterisk PBX-Konfiguration


  1. IT-Symposium 2005 7. April 2005 VoIP mit Asterisk PBX Voice over IP Telephonie mit Asterisk Nebenstellenanlagen John R. Covert DECUS-München Symposium 2005 3E06 VoIP � Technologiesituation � Netzprotokolle � Asterisk PBX-Konfiguration � Soft/ATA Klient-Konfiguration � Demo www.decus.de 1

  2. IT-Symposium 2005 7. April 2005 Technologiesituation � VoIP am Anfang: intern, auf privater Infrastruktur � Jetzt: im öffentlichen Internet � Hunderte von Providern � PC <--> PC, PC <--> PSTN � ATA (analog telephone adapter) Softphone � Telephonieren vom PC � Viele Produkte, auch kostenlose X-Lite SJphone www.decus.de 2

  3. IT-Symposium 2005 7. April 2005 Analog Adapter � Cisco ATA-186 � Sipura (tiptel in Deutschland) � HandyTone � FRITZ!box � GrandStream (VoIP Telephone) Voice over IP auf dem Vor mar sch � Jeder siebte Deutsche w ill per Inter net telefonier en � Schon acht von zehn Deutschen kennen Inter net- Telefonie � 44 Pr ozent begeister t von Einsatzmöglichkeiten � 14 Pr ozent planen Einstieg binnen zwölf Monaten � Niedr ige Kosten und Zusatzfunktionen über zeugen � Neue Studie von Roland Ber ger Mar ket Resear ch (VoIP News 4.4.2005) � � Gute Infoquelle: http:/ / www.ip- phone- for um.de/ www.decus.de 3

  4. IT-Symposium 2005 7. April 2005 Anbieter für Deutschland � AdvanceCall � GMX NetPhone � babble.net � Nikotel � BlueSIP � PURtel � Br oadnet � Sipgate � dus.net � Sipsnip � WEB.DE Fr eePhone � fr eenet iPhone Sipgate.de � Kostenlose verteilung geographischer bzw 01801 Nummer � Portierung ca. 4. Quartal 2005 � Softphone, Analog Adapter, Asterisk � Kostenfrei zu anderen VoIP Numm er � Verkauft ggf Analog Adapter � Pre-pay � Billige Minutenpreise www.decus.de 4

  5. IT-Symposium 2005 7. April 2005 Einstieg � Softphone dow nloaden, ggf. ATA � Bei einem Anbieter kostenlos anmelden (D-PSTN Rufnummer) � Kostenlose Seattle(Umgebung) Rfnr � Ankommende Gespräche kostenfrei � Gespräche an VoIP-Teilnehmer auch � Konto aufladen: ins PST N anrufen Bequemer Telefonieren � ATA oder VoIP kaufen � Normale Telefone anschließen � ggf VoIP Nebenstellenanlage (OpenSource Asterisk) einsetzen � Bei mehreren Providern anmelden � Least-cost Routing in Asterisk www.decus.de 5

  6. IT-Symposium 2005 7. April 2005 Kostenlos in die USA � Kostenlose SIP-Adresse (Provider) � Kostenlose 360 Area Code N r bei IPKall.com auf die SIP-Adresse � Amazon Yellow Pages (yp.a9.com) • Firma in den Yellow Pages suchen • 360-Nummer klingelt hier in Deutschland • Verbindung geht dann an die Firma • Hilton Disney Wor ld ht t p:// w w w .amazon.com/ gp/yp/ B0004UDK2G/ 002- 8827715- 4872835 Netw erkprotokolle � SIP - Session Initiation Protocol RFC 3261 � RTP/RTCP - Protocol for Real-Time RFC 1889 � STUN - Simple Traversal UDP N AT RFC 3489 � Auch H.323, IAX, RFC2833(DTMF), ... www.decus.de 6

  7. IT-Symposium 2005 7. April 2005 SIP - Session Initiation Anruf machen: -> INVITE <- SIP 407 Proxy-Authentica tion-Required -> ACK -> INVITE mit Authentifizierung (MD5) <- SIP 100 Trying <- SIP 183 Session Progress <- SIP 200 OK -> ACK SIP - Details (1 ->) INVITE sip:080 0225 5288 @ sip.covert.org SIP/2.0 Contac t: <sip:x25@ 6 2.8.199 .106:506 0> Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 Content-Type: applica tion/sdp From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 CSeq: 1 INVITE Max -Forw a rds: 70 To: <sip:0800 2255 288 @ sip.covert.org> Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 131 c9b1 4253 cb04 297 413fe 000 0000 7 User-Agent: SJLabs-SJphone /1.30.25 6b Content-Length: 366 www.decus.de 7

  8. IT-Symposium 2005 7. April 2005 SIP - Details (2 <-) SIP/2.0 4 07 Proxy Authentication Required Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 131 c9b1 4253 cb04 297 413fe 000 0000 7;rece ived=62.8.1 99.106 From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 To: <sip:0800 2255 288 @ sip.covert.org>;tag=a s28f7f493 Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow : INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contac t: Proxy-Authe nticate : Digest rea lm="asterisk", nonce ="2ed9 f3e8 " Content-Length: 0 SIP - Details (3 ->) ACK sip:080022 5528 8@ sip.covert.org SIP/2 .0 Content-Length: 0 Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 Max -Forw a rds: 70 CSeq: 1 ACK From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 To: <sip:0800 2255 288 @ sip.covert.org>;tag=a s28f7f493 Via: SIP/2.0/UDP 192 .168.2.1;bra nch=z9hG4 bKc0 a802 0101 31c9 b142 53c b042 9741 3fe0 000 0 007 www.decus.de 8

  9. IT-Symposium 2005 7. April 2005 SIP - Details (4.0 ->) INVITE sip:080 0225 5288 @ sip.covert.org SIP/2.0 Content-Length: 366 Contac t: <sip:x25@ 6 2.8.199 .106:506 0> Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 Content-Type: applica tion/sdp From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 CSeq: 2 INVITE Max -Forw a rds: 70 To: <sip:0800 2255 288 @ sip.covert.org> Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 000 0018 4253 cb04 03c c9df5 000 0000 9 User-Agent: SJLabs-SJphone /1.30.25 6b Proxy-Authorization: Digest userna me="x25 ",realm="asterisk",nonc e="2ed9f3e 8",uri="sip:08002 255 28 8@ sip.covert.org",response="7fe1 983 6112 2847 084e 888 0081 7a85 c3" SIP - Details (4.1 ->) v=0 o=- 332 1776 515 3 3217 7651 5 IN IP4 6 2.8.199.1 06 s=SJphone c=IN I P4 62.8.19 9.106 t=0 0 a=dire ction:active m=audio 1638 4 RTP/AVP 0 8 3 97 98 1 10 101 a=rtpm ap:0 PCMU/8 000 a=rtpm ap:8 PCMA/80 00 a=rtpm ap:3 GSM/8 000 a=rtpm ap:97 iLBC/80 00 a=rtpm ap:98 iLBC/80 00 a=fmtp:98 mode=20 a=rtpm ap:110 speex/8 000 a=rtpm ap:101 te lephone-event/8000 a=fmtp:101 0-1 1,16 www.decus.de 9

  10. IT-Symposium 2005 7. April 2005 SIP - Details (5 <-) SIP/2.0 1 00 Trying Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 000 0018 4253 cb04 03c c9df5 000 0000 9;rece ived=62.8.1 99.106 From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 To: <sip:0800 2255 288 @ sip.covert.org>;tag=a s671c1 c5f Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow : INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contac t: <sip:08002 2552 88@ 2 4.63.81 .3> Content-Length: 0 SIP - Details (6.0 <-) SIP/2.0 1 83 Session Progress Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 000 0018 4253 cb04 03c c9df5 000 0000 9;rece ived=62.8.1 99.106 From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 To: <sip:0800 2255 288 @ sip.covert.org>;tag=a s671c1 c5f Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow : INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contac t: <sip:08002 2552 88@ 2 4.63.81 .3> Content-Type: applica tion/sdp Content-Length: 207 www.decus.de 10

  11. IT-Symposium 2005 7. April 2005 SIP - Details (6.1 <-) v=0 o=root 3427 3 427 I N IP4 24.63 .81.3 s=session c=IN I P4 24.63 .81.3 t=0 0 m=audio 2560 4 RTP/AVP 0 8 101 a=rtpm ap:0 PCMU/8 000 a=rtpm ap:8 PCMA/80 00 a=rtpm ap:101 te lephone-event/8000 a=fmtp:101 0-1 6 (n.b. Asterisk kann noch mehr Audioformat e) SIP - Details (7 <-) SIP/2.0 2 00 OK Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 000 0018 4253 cb04 03c c9df5 000 0000 9;rece ived=62.8.1 99.106 From: "John Covert"<sip:x25@ sip.covert.org>;tag=39 1186 259 1491 7931 25 To: <sip:0800 2255 288 @ sip.covert.org>;tag=a s671c1 c5f Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow : INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contac t: <sip:08002 2552 88@ 2 4.63.81 .3> Content-Type: applica tion/sdp Content-Length: 207 [Conte nts sind dem vorigen Nachric hten gle ich] www.decus.de 11

  12. IT-Symposium 2005 7. April 2005 SIP - Details (8 ->) ACK sip:080022 5528 8@ 2 4.63.81 .3 SIP/2.0 Content-Length: 0 Contac t: <sip:x25@ 6 2.8.199 .106:506 0> Call-ID: E0AF5 2A6-1DD1-11B2-9 801-F6 0C9CB026 9B@ 192 .168.2.1 Max -Forw a rds: 70 CSeq: 2 ACK From: <sip:x25 @ sip.covert.org>;tag=3 9118 625 9149 1793 125 To: <sip:0800 2255 288 @ sip.covert.org>;tag=a s671c1 c5f User-Agent: SJLabs-SJphone /1.30.25 6b Via: SIP/2.0/UDP 192 .168.2.1;rport;branc h=z9hG4bKc 0a80 2010 131 c9b1 4253 cb08 6e9 641c 5 000 0000 d Und jetzt laufen also die RTP-Paketen. SIP - Weitere Funktionen � Anruf beenden • -> BYE • <- OK � Registrieren • -> REGISTER • <- SIP Trying • <- SIP 407 Proxy-Authentication... www.decus.de 12

  13. IT-Symposium 2005 7. April 2005 SIP - Weiteres � Weiterleiten • REFER (RFC 3515) � Ausloggen: • REGISTER m it Expires: 0 � Informieren • OPTIONS RTP/RTCP � Audio-Paketen • PCMU (µlaw ) • PCMA (alaw ) • GSM • SPEEX • iLBC • ... www.decus.de 13

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