1 / 134

Vovida Open Communication Application Library VOCAL

Vovida Open Communication Application Library VOCAL. System Architecture. What Is in This Module. Module Title: VOCAL System Architecture. Objectives: At the end of this module, you will be able to: Describe the VOCAL system architecture Describe the functionality offered by VOCAL

ebony-donovan
Télécharger la présentation

Vovida Open Communication Application Library VOCAL

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Vovida Open Communication Application LibraryVOCAL System Architecture

  2. What Is in This Module Module Title: VOCAL System Architecture Objectives: At the end of this module, you will be able to: • Describe the VOCAL system architecture • Describe the functionality offered by VOCAL • Describe the components of the VOCAL system and how they interact • Understand SIP call flows Module Length: 131 slides

  3. VOCAL Overview

  4. VOCAL – What is it? • Vovida Open Communication Library (VOCAL): • An open source, IP centric communication software, development platform and library. • It runs on: • Linux and Solaris operating systems. • Intel (I86) based hardware.

  5. VOCAL – What It Offers • VOCAL provides: Feature and Application Creation Operation System Support SIP Based Call Control and Switching

  6. SIP Based Call Control & Switching • VOCAL offers SIP based call control and switching: • User registration. • Call initiation. • Call modification. • Call termination.

  7. Operation Support Services • The operating support services within VOCAL provide the capabilities to: • Provision or configure the VOCAL system from a web GUI. • Monitor network elements from an SNMP network manager. • Add and manage subscribers and their feature subscriptions. • Authenticate subscribers. • Track billing information.

  8. Feature and Application Platform • VOCAL provides: • Basic features such as call forward, call blocking, call transfer, and call waiting. • A software library for new feature and application creation in: • C++. • Call processing language (CPL). • Java telephony API (JTAPI).

  9. VOCAL System Architecture

  10. PSTN Gateway 3rd Party Billing System Marshal Server Marshal Server Marshal Server Marshal Server Marshal Server Heartbeat Server Provisioning Server(s) Redirect Server(s) Feature Server(s) CDR Server(s) Policy Server(s) Internet Clearing House RADIUS H.323/SIP Translator MGCP/SIP Translator SNMP Network Manager MGCP Device SIP IP Phone H.323 Terminal VOCAL Architecture

  11. 2.INVITE 3.302 5. INVITE 6.302 4.INVITE 8. 180 (RING) 7. INVITE Redirect Server Marshal Server A Marshal Server B 1.INVITE 9. 200 (OK) Audio over RTP Channels 10. ACK SIP Phone User B SIP Phone User A A Basic SIP Call Using the VOCAL System (1) • 1.User A dials user B’s number. User A’s SIP phone sends an INVITE to marshal server. • 2.Marshal server A forwards the INVITE to the redirect server. • 3.The redirect server responds with a 302 containing information for marshal server A to contact marshal server B. • 4.Marshal server A forwards an INVITE to marshal server B. • 5.Marshal server B forwards the INVITE to the redirect server.

  12. 2.INVITE 3.302 5. INVITE 6.302 4.INVITE 8. 180 (RING) 7. INVITE Redirect Server Marshal Server A Marshal Server B 1.INVITE 9. 200 (OK) Audio over RTP Channels 10. ACK SIP Phone User B SIP Phone User A A Basic SIP Call Using the VOCAL System (2) • 6.The redirect server responds with a 302 containing information for marshal server B to contact user B. • 7.Marshal server B sends a INVITE to user B. • 8.User B’s SIP phone rings. The 180 message is sent back to user A’s SIP phone. • 9.When user B picks up the SIP phone, a 200 (OK) message is sent. • 10.User A’s SIP phone responds with an ACK message. • 11.The RTP path is now established.

  13. VOCAL Components

  14. User Agent

  15. PC with softphone application Komodo ATA 182/186 User Agent • VOCAL supports SIP user agents including: Pingtel xpressa Cisco 7960 SIP IP Phone

  16. Linux Workstation with: • Quicknet card • Vocal User Agent Linux Workstation with: • Quicknet card • VOCAL User Agent Analog Phone Analog Phone VOCAL User Agent • The VOCAL SIP user agent supports: • Call establishment. • Call waiting. • Transfer. • Registration with a marshal server or SIP proxy server.

  17. Marshal Server

  18. Marshal Servers • The Marshal Servers: • Are the only point of contact for all external devices. • Provides the logical function of the SIP proxy server and SIP registration server. • Performs one or more of these functions: • SIP message translation. • Authentication and security. • Billing.

  19. SIP Gateway Marshal Server Marshal Server SIP Phone User SIP Message Translation • The Marshal Server: • Checks • Translates • Logs • all SIP messages it receives from external entities. VOCAL System

  20. Marshal Functionality - Authentication • The Marshal Server supports these authentication methods: • No authentication. • Access control authentication – verification of IP address. • HTTP Digest authentication – verification of username and password.

  21. Provisioning Server RETRIEVAL REGISTER LOOKUP REGISTER 200 200 Redirect Server Marshal Server SIP Phone User No Authentication SIP Messages: REGISTER – Registers the address listed in the To header field 200 – OK

  22. The Marshal Server, verifies the IP address. Provisioning Server RETRIEVAL REGISTER LOOKUP REGISTER 200 200 Redirect Server Marshal Server SIP Phone User Access List Authentication SIP Messages: REGISTER – Registers the address listed in the To header field 200 - OK

  23. Provisioning Server RETRIEVAL REGISTER REGISTER LOOKUP REGISTER 401 200 200 Redirect Server Marshal Server SIP Phone User HTTP Digest Authentication SIP Messages: REGISTER – Registers the address listed in the To header field 200 – OK 401- Unauthorized

  24. Marshal Server CDR Server SIP Phone - Calling Party SIP Phone User - Called Party Marshal Functionality - Billing • Each Marshal Server sends the start and stop time of a call to the CDR Server. • The CDR Server forwards the data to 3rd party billing systems using the RADIUS accounting protocol. Marshal Server RADIUS 3rd Party Billing System

  25. Types of Marshal Server

  26. Gateway Internet PSTN Internetwork Marshal Conference Bridge Marshal Conference Bridge H.323 Terminal MGCP Endpoint Router SIP Phone Types of Marshal Servers H.323 Translator User Agent Marshal Gateway Marshal MGCP Translator

  27. Residential Gateway H.323 Terminal MGCP Endpoint SIP Phone User Agent Marshal • The User Agent Marshal Server: • Interact with User Agents. • Receives INVITE messages from User Agent. • Authenticates the user (against a user profile stored in a master file in the Redirect Server). • Requests routing information from the Redirect Server. H.323 Translator User Agent Marshal MGCP Translator

  28. SIP Gateway or SIP proxy PSTN Gateway Marshal SIP ISDN Gateway Marshal • Gateway Marshal Servers interact with SIP gateways or SIP proxy servers. • Gateways provide translation or interconnection between the IP and the PSTN network.

  29. Conferencing • The VOCAL system supports two types of conferencing: • Meet-Me – users call a predefined number at predefined time. • Ad-Hoc – user adds multiple users to a call. • Ad-Hoc conferencing requires a Conference Bridge Marshal.

  30. SIP Gateway PSTN Marshal Server Conference Bridge or Multimedia Conference Unit (MCU) Gateway Marshal Server Analog Phone C SIP Phone B SIP Phone A Meet-Me Conferencing INVITE /200 /ACK • Meet-Me conferencing allows any users to call a conference bridge number. • RTP media channel is established for each user. • The conference bridge mixes the audio streams.

  31. User C User A User B What is Ad-Hoc Conferencing? • With ad-hoc conferencing a user adds multiple participants to a call: • User A and User B are in a call. User A wishes to add User C to the call. • User A places User B on hold and calls User C. • User C answers. • User A adds User C to the call with User B. The call is now a conference call. 1. A puts B on hold 2. A calls C 3. User A, B, and C are in a conference call.

  32. Conference Bridge INVITE (HOLD) TRANSFER INVITE/180 200/ACK SIP Phone B SIP Phone C SIP Phone A 200/BYE/OK How would ad-hoc conferencing work with SIP? User A is talking to User B User A puts User B on HOLD RTP User A calls User C INVITE/180/200/ACK User A transfers User C to conference bridge RTP User A transfers User B to conference bridge RTP TRANSFER INVITE/180/200/ACK RTP 200/BYE/OK INVITE/180/200/ACK RTP

  33. Implementation Issues • The previous call flow diagram illustrates an ideal implementation. There are these implementation issues: • Most conference bridges do not use SIP. • Therefore a SIP gateway is required. • However, most SIP gateway cannot handle multiple calls with the same call ID. • All conference calls use the same call ID. • At the time of implementation, there was no SIP standard on conferencing.

  34. SIP Gateway Marshal Server Conference Bridge or Multimedia Conference Unit (MCU) Conference Bridge Marshal Server SIP Phone C SIP Phone B SIP Phone A VOCAL Solution – Conference Bridge Marshal Server Insert common call ID in outbound SIP messages Removes common call ID in inbound SIP messages • The Conference Bridge Marshal Server: • Inserts a common call ID for outbound SIP messages to the SIP gateway. • Removes the common call ID and inserts unique call IDs for inbound SIP messages from the SIP gateway.

  35. Internetwork Marshal • The Internetwork Marshal Server is used to interconnect with: • Other SIP systems that use the OSP protocol. • Clearinghouses.

  36. Translators

  37. SIP Gateway Marshal Server Marshal Server Marshal Server Marshal Server H.323 Terminal MGCP Call Agent SIP Phone User Translator Functionality • VOCAL is SIP based. • Supports non-SIP endpoints using translators. • Supports H.323 and MGCP endpoints. VOCAL System H.323 Translator MGCP Translator

  38. H.323 Translator • Provides call signaling translation between H.323 endpoint and SIP server. • H.323 endpoints appear as SIP user agents to the VOCAL system. • The current implementation works with Microsoft NetMeeting 3.01 as the H.323 endpoint.

  39. MGCP Translator • Provides call signaling translation between MGCP endpoint and SIP server. • MCGP translator can act like a MGCP call agent that controls MGCP gateways.

  40. Provisioning Server

  41. Provisioning Server • The Provisioning Server: • Stores data on all users and servers within the VOCAL system. • Accessible from a Java-based GUI via an Internet browser.

  42. Provisioning GUI • The Provisioning GUI is used to: • Configure the VOCAL system. • Administer users and enable user’s features. • Subscribe or unsubscribe user’s features.

  43. Technician Screen • The Technician screens allows you to configure or provisioning the VOCAL servers.

  44. Administrator Screen • The Administrator screen allows you to add users and enable their features.

  45. User Screen • The User’s screen allows you set the user’s features. • Note – the user’s features must first be enabled by the administrator before a user can set it.

  46. Marshal Server Redirect Server Feature Server Provisioning Server - Data Storage All data is stored in the Provisioning Server as XML files in this directory: /usr/local/vocal/provisioning_data Provisioning Server Provisioning GUI Subscribe Notify Method

  47. Redirect Server

  48. Redirect Server • The Redirect Server provides these SIP services and functions: • Registration. • Redirection. • Location. • The Redirect Server provides routing information to the Feature and Marshal Servers to route a call.

  49. 5. INVITE 6. 302 2.INVITE Marshal Server B Redirect Server Marshal Server A 3. 302 4.INVITE 8. 180 (RING) 1.INVITE 7. INVITE 9. 200 (OK) 10. ACK SIP Phone User B SIP Phone User A Audio over RTP Channels Review - A Basic Call involving Marshal and Redirect Servers • Marshal Servers forwards INVITE messages to the Redirect Server to obtain routing information. • The Redirect Server responds with a 302 message containing the routing information.

  50. How the Redirect Server Determines Route • The Redirect Server determines route by: • Retrieving a previously built subscriber list or the dial plan. • Building a contact list from information in the INVITE message. • Generating a 302 message with the routing information.

More Related