VoIP - Cisco CME & IP Communicator

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VoIP - Cisco CME & IP Communicator

  1. 1. OVERVIEW
  2. 2. CALL SCENARIO
  3. 3. THE CISCO Call Manager Express <ul><li>OPERATION: </li></ul><ul><li>IT tracks all active VOIP | POTS components like phones , gateways , bridges and trans-coding resources </li></ul><ul><li>Configure CLI commands for Cisco IP communicator using the SCCP </li></ul><ul><li>Configure CLI Dial-peer commands for Sjphone using the SIP </li></ul><ul><li>Configure CLI Dial-peer commands for e911 services using POTS </li></ul><ul><li>Provides end to end connectivity between any soft phone using either SCCP or SIP </li></ul>
  4. 4. THE CISCO Call Manager Express <ul><li>THE BASIC’s of VOIP DIAL-PEERS: </li></ul><ul><li>Router(config)#  dial-peer voice number voip </li></ul><ul><li>Router(config-dial-peer)# destination-pattern string </li></ul><ul><li>Router(config-dial-peer)# session target { ipv4: destination-address | dns:[] host-name } </li></ul><ul><li>Router(config-dial-peer)# dtmf-relay [ cisco-rtp ] [ h245-signal ] [ h245-alphanumeric ] </li></ul><ul><li>Router(config-dial-peer)# session protocol { sipv1 |sipv2 |cisco} </li></ul><ul><li>Dual tone multi frequency relay is the mechanism where a local VOIP gateway listens for DTMF digits during a call and then sends them uncompressed as either RTP or H.245 packets to the remote VOIP gateway which regenerates DTMF digits and prevents digit loss due to compression </li></ul><ul><li>This in-band uses a special payload type identifier [PTI] in the RTP header of the voice media stream to distinguish digits from the DTMF pad from actual voice communication </li></ul>
  5. 5. THE CISCO Call Manager Express <ul><li>THE BASIC’s of IOS TELEPHONEY : </li></ul><ul><li>DHCP service </li></ul><ul><li>IP source address </li></ul><ul><li>SKINNY communication port </li></ul><ul><li>Number of IP phones [max 8] </li></ul><ul><li>Dual-line extensions </li></ul><ul><li>Phone language </li></ul><ul><li>Call Progress Tone </li></ul><ul><li>First extension number </li></ul><ul><li>Direct Inward Dial </li></ul><ul><li>Call forward voice mail service </li></ul><ul><li>THE BASIC’s of POTS DIAL-PEERS: </li></ul><ul><li>Router(config)# dial-peer </li></ul><ul><li>voice number pots </li></ul><ul><li>Router(config-dial-peer)# </li></ul><ul><li>destination-pattern string </li></ul><ul><li>Router(config-dial-peer)#  port </li></ul><ul><li>location </li></ul>
  6. 6. THE CISCO IP COMMUNICATOR <ul><li>The IP phone acts as a dumb terminal while the CME is responsible for the entire setup and the tear down of the call. </li></ul><ul><li>The IP phone registers its name, device type and IP address with the CME and then provides its IP port number on which it will receive and send media messages . </li></ul><ul><li>The CME assigns the soft-keys corresponding to each event that takes place during the call scenario like off hook, new call, redial, hold, etc. </li></ul><ul><li>On occurrence of any of the events the CME sends soft-key event messages to the phone and sends periodic call status info messages. </li></ul>
  7. 7. THE SJ SIP SOFT - PHONE <ul><li>SJ phone is a SIP soft-phone which does not require to be registered with a server if it is being accessed within the LAN [ it is its own user-agent client and user-agent server] </li></ul><ul><li>Integrates with the Microsoft Loopback Driver to receive, parse and translate SIP messages and represent the SIP call scenario </li></ul><ul><li>Supports 1. SYMBIAN OS </li></ul><ul><ul><ul><ul><li>2. MAC OS </li></ul></ul></ul></ul><ul><ul><ul><ul><li>3. WINDOWS OS </li></ul></ul></ul></ul>
  8. 8. SKINNY – SIP CALL SETUP AND TEARDOWN
  9. 9. SKINNY – SIP CALL SETUP AND TEARDOWN
  10. 10. SKINNY – SIP CALL SETUP AND TEARDOWN
  11. 11. SKINNY – SIP CALL SETUP AND TEARDOWN
  12. 12. Part- II: Development of basic SoftPhone <ul><li>Functionality: </li></ul><ul><li>Basic SIP softphone client (like k-lite,sjphone etc) </li></ul><ul><li>Can Register with any SIP server </li></ul><ul><li>Can receive/make calls from/to SIP enabled phones. </li></ul><ul><li>Audio codec used: G.729 </li></ul><ul><li>Can add/modify program to achieve more functionality like call forwarding, conferencing, video chat and many more. </li></ul><ul><li>Programming Facts for this project: </li></ul><ul><li>Operating System: Win Vista </li></ul><ul><li>portSIP SDK for Microsoft .NET </li></ul><ul><li>Programming Language: C# </li></ul><ul><li>IDE: Microsoft Visual Studio 2008 </li></ul><ul><li>Couldn’t have been really possible without help of portSIP SDK documentation. </li></ul>
  13. 13. PortSIP SDK for Microsoft .NET <ul><li>Basic Features: </li></ul><ul><li>Support platforms: Windows 2000/XP/2003/Vista, Windows Mobile 5/6, Nokia S60 3rd FP2. </li></ul><ul><li>Support servers: Cisco CallManager, Open SER,  SER, Asterisk, Portaone, Radvision, Nortel, Rainbow, Avaya and other SIP Platforms. </li></ul><ul><li>Audio call: G.711 aLaw/uLaw, GSM, iLBC, G723.1, G729. </li></ul><ul><li>Video call: H263, H263-1998, H264. </li></ul><ul><li>Call transfer: Attended transfer, Blind transfer. Call forwading, Call hold, mute speaker, mute microphone </li></ul><ul><li>IM Support: SIMPLE(Presence, Subscribe, Pager message) and XMPP. </li></ul><ul><li>For full features please see http://www.portsip.com/features.htm </li></ul><ul><li>Architecture: </li></ul><ul><li>SIP SDK for Microsoft.Net. It is easy to use with c# language once it is understood. </li></ul><ul><li>Contains 3 main libraries </li></ul><ul><ul><li>DeviceManagerLibV4 </li></ul></ul><ul><ul><li>PortSIPCoreLibV4 </li></ul></ul><ul><ul><li>PortXMPPLibV4 </li></ul></ul><ul><li>DeviceManagerLibV: </li></ul><ul><ul><li>Its easy to select audio/video devices on a computer for VOIP communication, using various functions which are written in this library </li></ul></ul><ul><li>PortSIPCoreLibV4 </li></ul><ul><ul><li>This library implements the core Session Initiation Protocol stack. </li></ul></ul><ul><ul><li>It has rich set of functions and events for SIP . </li></ul></ul><ul><li>PortXMPPLibV4 </li></ul><ul><ul><li>It has implementation of XMPP protocol for Instant Messaging. </li></ul></ul>
  14. 14. Soft-phone Client <ul><li>Welcome Screen: </li></ul><ul><li>You need to enter your username, password, proxy server name, domain name of proxy server, proxy server port number (If any). </li></ul><ul><li>You must have account for any SIP server to use this softphone. </li></ul><ul><li>We used www.voxalot.com username/password for our testing purpose. </li></ul><ul><li>If username/password is correct, it will try to register it on SIP proxy server you provided. </li></ul><ul><li>If registration is successful, you can make/receive calls from SIP enabled devices. </li></ul><ul><li>To authentication and registration we used initialize() and registerServer() functions from portSIP SDK. </li></ul>
  15. 15. Softphone Client: cntd <ul><li>Phone Screen: </li></ul><ul><li>Once you are successful authenticated and registered, you will see this screen. </li></ul><ul><li>You can make call or received calls from sip enabled devices now. </li></ul><ul><li>Here we are trying to call ‘408334’ which is also a registered number of voxalot. </li></ul><ul><li>You can see the log of the various events occuring. </li></ul><ul><li>Same way it will give you notification for incoming call. </li></ul><ul><li>We have used following main functions and events from portSIP SDK to make this achieved. </li></ul><ul><li>Methods </li></ul><ul><ul><li>Call(ref string callTo,hasSDP) : To place a call. Returns sessionid </li></ul></ul><ul><ul><li>answerCall(int sessionID) : To answer an incoming call. </li></ul></ul><ul><ul><li>rejectCall() and terminateCall() : To reject/end a call </li></ul></ul><ul><li>Events: </li></ul><ul><ul><li>inviteTrying() : when call is trying </li></ul></ul><ul><ul><li>inviteRinging() when phone is ringing </li></ul></ul><ul><ul><li>inviteAnswered() when phone is answered by a person </li></ul></ul><ul><ul><li>inviteIncoming() when there is an incoming call </li></ul></ul><ul><ul><li>inviteClosed(() when person disconnects the call </li></ul></ul>
  16. 16. REFRENCES & AREA OF CONTRIBUTION <ul><li>http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/dialplan.html </li></ul><ul><li>http://www-europe.cisco.com/univercd/cc/td/doc/product/software/ios122/122sup/122csum/csum3/122cvvf/vsf_r.htm#1736728 </li></ul><ul><li>http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srstsa.html </li></ul><ul><li>www.portsip.com </li></ul><ul><li>CHINMAY PADHYE </li></ul><ul><li>CME – IP COMM. – SJ PHONE SIMULATION </li></ul><ul><li>NEHA SHARMA </li></ul><ul><li>SKINNY – SIP CALL FLOW ANALYSIS </li></ul><ul><li>VAIBHAV KULKARNI </li></ul><ul><li>DEVELOPMENT OF SIP SOFT-PHONE </li></ul>

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