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  • 1. The design, comparison and current status of TWAREN voicemail systems Chia-Hung Hsu Li-Chi Ku Che-Nan Yang National Center for High-performance National Center for High-performance National Center for High-performance Computing Computing Computing 7, R&D Rd. VI, Hsinchu Science Park 7, R&D Rd. VI, Hsinchu Science Park 7, R&D Rd. VI, Hsinchu Science Park Hsinchu 300, Taiwan Hsinchu 300, Taiwan Hsinchu 300, Taiwan 886-3-5776085 ext. 385 886-3-5776085 ext. 208 886-3-5776085 ext. 224 chhsu@nchc.org.tw lku@nchc.org.tw yangcn@nchc.org.tw ABSTRACT headquarter to provide SIP to SIP, SIP to PSTN, NAT TaiWan Advanced Research & Education Network (TWAREN) transversal, voice meeting, and voice mail services to the users. is eager to bring integrated VoIP technologies to its academic The user base is mainly TWAREN members at this time. It will and research community. Among those flourished Session be extended to faculties and students in colleges as the next step, Initiation Protocol (SIP) based solutions, we have established a and finally be broadened to pan-Asia countries. SIP system comprised mainly iptel’s free software SER. An The voice mail service currently deployed on TWAREN consists attempt of bringing in more integrated VoIP technologies, of three different solutions: SEMS[9], Unity[6] and Asterisk [1] including the voice mail systems from SEMS, Cisco Unity and [2]. SIP Express Media Server (SEMS) is a SIP-based media Asterisk, has been made. The comparison of the three voice mail server. After integrated with SIP Express Rotuer (SER) and systems and the current status of the VoIP service deployment necessary plugin modules, it becomes a simple but functional on TWAREN will be discussed in this paper. voice mail system. Cisco Unity is a component of Cisco Unified Communication Systems. As a commercial product, it combines Keywords the functionality of E-mail, voice chat, FAX and voice mail. TWAREN, VoIP, SEMS, Unity, Asterisk Finally, Asterisk is a open source telephone exchange software officially supported by Digium. Besides the complete functionality of VoIP, it supports multiple transmission 1. INTRODUCTION protocols as well as abundant telephone hardware interfaces. VoIP is the hottest topic of Internet application nowadays. Due to its operating on top of the solid foundation of IP network, This article will provide the idea behind the voice mail services VoIP significantly reduces the cost among distributed campuses deployed on TWAREN. All three voice mail solutions will be and branches. compared and discussed. TaiWan Advanced Research & Education Network (TWAREN) VoIP service[11] has been established and running for one year. The structure of the service is shown in figure 1. Figure 2 TWAREN VoIP entrance webpage In 2. TWAREN VoIP Services Figure 1 TWAREN VoIP architechture 2.1 The VoIP Services Deployment The original SIP service of TWAREN has been introduced in current configuration, Cisco 1721s has been installed in each of our previous article[5]. In the 2006, some underlying structures the four core nodes (Taipei, Hsinchu, Taichun, Tainan) of has been modified to include the voice mail support. Figure 2 TWAREN. This equipment connects the VoIP services of shows the entrance web page of the TWAREN VoIP integrated TWAREN gigapops as well as the PSTN via a BRI interface. A services. This page allows the users to basic settings to their Cisco 3725 serves as a VoIP gateway connecting those Cisco accounts. Administrators can remotely control the VoIP gateway 1721s and the PSTN with a PRI interface. The main SIP proxy and supporting servers are established in the Hsinchu
  • 2. through this web page as well as the VoIP gateway control Signaling Media interface (as shown in figure 3). Application Modules Session Dialog Audio Audio Handling Handling Layer Modules SER IPC SER IPC RTP File I/O Server Client Layer Figure 5 SEMS architechture Figure 6 shows the signal transduction sequence between the TWAREN SIP proxy server (SER) and the SEMS server. When SIP UA1 issued “INVITE SIPUA2@sipdomain” command to SIP proxy trying to initiate the voice session with SIP UA2, the SIP proxy noticed that SIP UA2 was not registered (online). Figure 3 TWAREN VoIP gateway control interface Instead of rejecting the invitation, SIP proxy will transfer this session to the SEMS server. SEMS replied a “200 OK” message, Besides the VoIP common exchange platform project[10] and the RTP voice session with SIP UA1 was established. SEMS running by TANET, TWAREN SIP network currently has VoIP continued to collect the voice meessage from SIP UA1, until the peerings with many universities and research institutions, “BYE” or “CANCEL” message was received. The collected including National Taiwan University, National Kaohsiung voice message was then sent to the E-mail box of the recipient University, Industrial Technology Research Institute, and through the SMTP server as an attachment. National Space Organization. National Taiwan University National Ilan University National Chung Cheng University SIP Proxy Server SEMS IP Network National Center for High -Performance Computing SIP UA1 Industrial Technology Research Institute Mail Server SIP UA2 National University of Kaohsiung (unregistered ) INVITE SIPUA2 National Sun Yat -sen University INVITE SIPUA2 National Space Organization 200 OK 200 OK RTP Session Figure 4 TWAREN VoIP peering organizations SMTP Voice Mail 2.2 SEMS SEMS is a SIP based media server. Through the SIP proxy and Figure 6 Signal transduction sequence of SEMS voice mail necessary plugin modules, it enables a Linux running machine into a IP telephony system. The default configuration contains The internal flow of messages inside SEMS is shown in figure modules capable of announcement, voice mail, conference, 7. When SIP proxy transferred the “INVITE” message to the ISDN gateway, interactive voice responses, echo and other SEMS server, the SER IPC Server was activated and the useful functions. The open architecture of this plugin system “amSession” was started. Suitable modules were loaded by gives the necessary freedom for administrators to design new SEMS preparing for the upcoming voice session. At the end of modules to fit their specific needs. The audio module layer the ”amSession”, SEMS responded a “200 OK” message, sended defines the supported codecs and file formats, which includes the SDP to the SER IPC client, and started the “onSessionStart” wav: G711u, G711a, wav file, gsm: GSM 06.10 codec, iLBC: session, which completed the RTP session establishment. iLBC internet low bitrate codec and mp3: mp3 files. The file Afterward, the SEMS collected the voice packet from SIP UA1, will be presented as the attachment in a notification E-mail to until “BYE” or “CANCEL” was received. the recipients. SEMS can communicate with SER through Unix socket or FIFO. Figure 8(a) shows an example config to instruct SEMS to communicate with SER through Unix socket. The ! lookup(“location”) line indicates the situation that the user is not registered to the SIP Proxy server. Figure 8(b) shows another example where the SEMS communicates with SER through FIFO.
  • 3. attachment inside the E-mail sent to the recipients. Figure 10 shows how the E-mail looked like. The recipient can listen to SER IPC Server amSession the voice message by simply clicking the attachment icon. Factory Container Voicemail IVR ... DialogState SER IPC Client amSession Figure 10 Voice mail example DialogState::onSessionStart 2.3 Unity Cisco Unity is a component of the commercial Cisco Unified Communication Systems which provides E-mail, voice, FAX Figure 7 SEMS call flows integrated service as well as a full-featured voice mail service. Cisco Unity supports Microsoft Exchange and IBM Domino as the backend SMTP server. It can also utilize the Dynamic Domain Name Servers, Directory Servers and Message Stores as shown in figure 11. Figure 8 The example of SER-SEMS communication (a) (upper) using Unix socket (b) (lower) using FIFO The current TWAREN VoIP services are illustrated as shown in figure 9. The announcement service was accomplished by recording an announcement wav file on the SEMS server. When SIP UA initiated a voice session with a predefined number, the SEMS server played this wav file back in the voice session so the SIP UA can receive them. Announcement RTP SIP UA1 SEMS Conference Figure 11 Unity system components RTP SIP UA1 2.3.1 Unity Voice Mail Service RTP Cisco Unity provides voice mail functionality to Cisco CallManager and tranditional PSTN telephone system. It also SIP UA2 SEMS features VPIM (Voice Profile for Internet Mail,RFC-3804) and Voicemail AMIS (Audio Messaging Interchange Specification) modules to support the message exchange to other industrial standard RTP SMTP systems. Besides, the SIP integrated function strengthened Cisco SIP UA1 Unity the power to support SIP Proxy, SIP client and SIP- enabled gateway so the voice mail service can be extended to all SEMS Mail Server SIP users. Figure 9 SEMS applications in TWAREN 2.3.2 Unified Communication Cisco Unity offers a complete messaging service including The conference service was also accomplished by pre-recording reading E-mails through telephone, checking the voice mail wav files in the SEMS server. When multiple users dialed the from the Internet as well as sending FAX to other FAX machine same conference room number, each of them received the pre- through a FAX server. The centralized Message Store facility recorded message and then the conference meeting began. In the allows the stored messages to be retrived as different formats. case of the voice mail service, SEMS started collecting voice messages after firstly sending recipients a predefined voice To show the complete potential of Cisco Unity, TWAREN instruction. The voice messages was then packaged as an installed all these supported components as shown in the table 1.
  • 4. Table 1. Installed components on TWAREN To summarize the capability of the Unity[4] currently deployed on TWAREN, the platform model is MCS-7825-H2-ECS1 with Category Components maximun user number of 1000 and at most 12 concurrent text to speech sessions. The longest possible recording capability is Voice Mail Cisco Unity 67,271 minutes, which is about 1,121 hours. SIP Proxy iptel SER Dynamic Domain Microsoft DNS Name Server SIP Proxy Server Directory Server Active Directory Message Store Microsoft Exchange 2000 IP Network SIP UA1 Unity The signal transduction sequence of Cisco Unity voice mail is depicted in figure 12. The voice message from SIP UA3 was SIP UA2 (unregistered ) transferred to Unity Server by a SIP Proxy server. After the SIP INVITE SIPUA2 Proxy transferred the responses of the Unity Server back to SIP INVITE SIPUA 200 OK 2 UA3, the RTP session between SIP UA3 and the Unity server 200 OK was established. The voice messages of SIP UA3 were RTP Session Notify transferred to Microsoft Exchange Server and then E-mailed to INVITE SIPUA2 SIP UA1 through the SMTP protocol. INVITE SIPUA 200 OK 2 200 OK RTP Session Figure 13 Signal transduction sequence of Unity voice mail 2.4 Asterisk Asterisk is an open source PBX software officially supported by Digium. Developed by open source developers, it quickly became a mature and powerful solution supporting numerous codecs, file formats, protocols and hardware interfaces. Asterisk contains several components as shown in figure 14[7]. The PBX switching core accepts and dispatches calls receiving from the hardware interfaces. The application launcher processes the application level duties like ringing the phone and voice mail delivery. Codec translator accommodates the codecs using by different users and interfaces. Scheduler and I/O Manager manages the scheduling and the I/O process. Asterisk also supports many loadable modules and APIs to extend its functionality in every aspects. The Channel API serves the client-side channels. Application API processes the server-side application services. Codec translator API translates the data between differenct codecs. Finally the file format API reads and writes all supported file formats. Figure 12 Unity message flow in TWAREN Besides the way to send voice messages to user by E-mail attachments, Cisco Unity also supports another way to the voice message delivery. In this case, the voice data was stored on the Unity server for users to retrive by dialing a specific number and then listening to the stored message. The signal flow of this mode is shown in figure 13. SIP UA1 sent a "INVITE" to SIP UA2 and this message was silently redirected to the Unity server by the SIP Proxy. After the RTP session was established, the voice messages were continuously recorded until the appearance of the "BYE" or "CANCEL" message. Unity server left a notification on the SIP proxy so that once the SIP UA2 registered to the SIP Proxy server, it got notified. SIP UA2 can then dial a number to establish a RTP session with the Unity server to retrive the voice message.
  • 5. IAX (Inter-Asterisk Exchange) H.323 SIP MGCP SCCP (Cisco Skinny) The signal transduction flow of Asterisk voice mail is shown in figure 15. When SIP UA1 dialed SIP UA2 and the SIP proxy realized that SIP UA2 was not yet registered, the message was then transferred to Asterisk. A common user accounts pool has been pre-configured on the Asterisk system such that the call can be redirected to the voice mailbox corresponding to the SIP UA2. When accepting the incoming calls, the behavior of Asterisk is controlled by two configuration files: extensions.conf and voicemail.conf[3]. The extensions.conf determines the path each incoming call is redirected to, as well as the voice mail if the recipient is busy or not registered. The voicemail.conf Figure 14 The architechture of Asterisk components determines the file format used in the voice message recording and whether to send a simple notification E-mail to the recipient Table 2 shows the codec list currently supported by Asterisk. or an E-mail containing the voice message itself as an These codecs cover the usage of PSTN telephones, cell phones attachment. as well as all common VoIP software codecs. Asterisk also supports many industrial standard transmission protocols which makes it a perfect choice of a telephone exchange server. Table 2. Supported Codecs of Asterisk Supported Codecs ADPCM G.711 (A-law and µ-law) G.723.1 (pass through) G.726 G.729 GSM iLBC Linear LPC-10 Speex Asterisk server can directly connect to PSTN network by adding Figure 15 The signal transduction sequence of Asterisk voice T1/E1/J1 interface cards to the server (for example, Digium mail service Wildcard TE410P). Because the broadness of the codec and There are two ways to retrive the voice messages: one by protocol support, and the ability to directly PSTN connecting, checking the attachment in the notification E-mail, and another Asterisk can serve as a great exchange server between different by dialing into the voice mailbox. In current TWAREN VoIP VoIP softwares and telephone hardwares. The application of configuration, the VoIP connections are processed by the SIP Asterisk on the voice mail service will be the main focus of this Proxy server (SER). Asterisk only serves as a voice mail service article. provider, so a simple extensions.conf is used to redirected all the Table 3. Supported Protocols of Asterisk incoming calls to the Asterisk server to the voice mailbox. Supported Protocols 2.5 Comparison of the current three voice mail services
  • 6. The difference of the three voice mail services is compared and File format wav wav wav/gsm summarized in table 4. Among these three solutions, SEMS is easier to setup, less resource consuming. The voice messages Database Shared Standalone Standalone can be sent to the recipients by combining with a regular E-mail server. The constraint of this solution is that it is not so feature UI Friendly Low Medium Low rich and it needs a specific SIP Proxy server to be functional. Cisco Unity is a commercial software, therefore the cost of platform establishment and maintenance is much higher than the other two solutions. Due to its commercial property, it pays 4. REFERENCES more attention to the support of information security. Asterisk is [1] Asterisk, http://www.asterisk.org/, Last updated: a extremely feature rich software supporting a very complete set Jul/15/2006, Last viewed: Jul/31/2006. of codecs, protocols and file formats. As it is no doubt a great choice to be an exchange point to hetergenous VoIP solutions, it [2] Asterisk extensions.conf settings, is rather complicated and difficult to configure. http://www.digium.com/en/docs/asterisk_handbook/extensi ons.conf.html, Last viewed: Jul/31/2006. 3. Conclusions [3] Asterisk voicemail.conf settings, TWAREN VoIP services has been available for over one year. http://www.digium.com/en/docs/asterisk_handbook/voicem The usage history shows that most users using this service to ail.conf.html, Last viewed: Jul/31/2006. make PSTN calls and cross branches SIP phone calls. The voice [4] Cisco Unity Tools, mail services are provided to serve the nature that most http://www.ciscounitytools.com/Documents.htm TWAREN VoIP users do not stay near their SIP phone. By implementing SEMS, Unity and Asterisk, TWAREN VoIP users [5] Hsu C.H., Kuo S.H., Chang I.C., Yang C.N. The SIP-Based can have their voice messages by checking the attachment of E- VoIP deployment on TWAREN, TANET2005 Conference mail notifications or by dialing into the voice mail systems. Proceding, 15. Since SEMS and Asterisk are GPL licensed software, greatly [6] http://www.cisco.com/en/US/customer/products/sw/v minimize the cost of deployment, so TWAREN VoIP services oicesw/ will emphasize these two solutions to develop a more user friendly VoIP environment in the future. [7] Mark Spencer, Mack Allison, Christopher Rhodes, The Asterisk Documentation Team. “The Asterisk Handbook Table 4. The comparison among the three solutions V2”, http://www.digium.com/en/docs/asterisk_handbook/handbo SEMS Unity Asterisk ok-draft.pdf Develop / [8] Raphael Coeffic, Stefan Sayer, An Introduction to SEMS, iptel CISCO Digium Supported by the SIP Express Media Server, License GPL License GPL http://ftp.iptel.org/pub/sems/doc/current/Readme.html Hardware X86 [9] SIP Express Media Server, http://www.iptel.org/sems MCS Server X86 PC Requirement PC [10] TANET VoIP services, http://www.edu.tw/EDU_WEB/ Operating Linux/ Windows Linux/BSD/ EDU_MGT/MOECC/EDU0688001/net_phone/index. System BSD 2000 Mac OS htm [11] TWAREN VoIP entrance, http://ser.nchc.org.tw/ Cost of Having Low High Low Difficulty to Easy Medium Hard setup Difficulty to obtain Easy Hard Easy resources Method to SMTP/ SMTP/ retrive voice SMTP mail SIP SIP Sendm Microsoft Sendmail Mail Server ail Exchange Postfix Postfix Generate No Yes No reports Security Low High Low SIP/H.323/ Protocol SIP/ SIP IAX/Skinny/ supported Skinny PSTN

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