The design, comparison and current status of TWAREN
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
firstname.lastname@example.org email@example.com firstname.lastname@example.org
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, Unity and Asterisk 
including the voice mail systems from SEMS, Cisco Unity and . 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 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. 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
through this web page as well as the VoIP gateway control
interface (as shown in figure 3).
Session Dialog Audio Audio
Handling Handling Layer Modules
SER IPC SER IPC RTP
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 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
National Center for High -Performance Computing SIP UA1
Industrial Technology Research Institute
National University of Kaohsiung (unregistered )
National Sun Yat -sen University INVITE SIPUA2
National Space Organization 200 OK
Figure 4 TWAREN VoIP peering organizations SMTP
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.
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
attachment inside the E-mail sent to the recipients. Figure 10
shows how the E-mail looked like. The recipient can listen to
the voice message by simply clicking the attachment icon.
Voicemail IVR ...
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.
Conference Figure 11 Unity system components
2.3.1 Unity Voice Mail Service
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.
Table 1. Installed components on TWAREN To summarize the capability of the Unity 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
SIP Proxy Server
Directory Server Active Directory
Message Store Microsoft Exchange 2000 IP Network
The signal transduction sequence of Cisco Unity voice mail is
depicted in figure 12. The voice message from SIP UA3 was SIP UA2
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
UA3, the RTP session between SIP UA3 and the Unity server 200 OK
was established. The voice messages of SIP UA3 were RTP Session
transferred to Microsoft Exchange Server and then E-mailed to INVITE SIPUA2
SIP UA1 through the SMTP protocol. INVITE SIPUA
Figure 13 Signal transduction sequence of Unity voice mail
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. 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.
IAX (Inter-Asterisk Exchange)
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. 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
G.711 (A-law and µ-law)
G.723.1 (pass through)
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.
2.5 Comparison of the current three voice mail services
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
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minimize the cost of deployment, so TWAREN VoIP services oicesw/
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iptel CISCO Digium
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License GPL License GPL http://ftp.iptel.org/pub/sems/doc/current/Readme.html
Hardware X86  SIP Express Media Server, http://www.iptel.org/sems
MCS Server X86 PC
Requirement PC  TANET VoIP services, http://www.edu.tw/EDU_WEB/
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Cost of Having Low High Low
Easy Medium Hard
obtain Easy Hard Easy
Method to SMTP/ SMTP/
retrive voice SMTP
mail SIP SIP
Mail Server ail
No Yes No
Security Low High Low