The trend of the recent telecommunication network development is integrating
different networks. In Taiwan, a dual network project, also known as Integrated Beyond
3rd Generation, is one of the pioneer projects which integrates GSM (Global System
for Mobile Communications) and VoIP (Voice over IP). Therefore, many issues can be
investigated in this domain.
GSM network has evolved from 2nd generation to 2.5 generation GPRS (General
Packet Radio Service), and now to 3G (3rd generation) UMTS (Universal Mobile
Telecommunications System) version R4. But the 3G All-IP (version R5) network still not
mature in 2005 yet. Thus, a smooth interaction between GSM network and IP Telephony
still face a lot of difficulties. With the trend of fast growth of the IP phone users, the need to
develop such a smooth integration to meet the time to market is urgent.
The concept of using only one MSISDN (Mobile Station Integrated Services Digital
Network) number for each user among different wireless networks (including GSM and IP
networks) just fits the current trend of ever more complex networks but simpler human
operation method. In the future, a user can be present in different networks by using only
one personal number, i.e., the user can be reached by dialing the personal number no matter
what networks the user is on.
As VoIP service becomes more popular, more calls originated from Internet devices,
such as computer soft-phones and hardware IP phones, to the PSTN (Public Switched
Telephone Network) or PLMN (Public Land Mobile Network). However, calls originated
from the PSTN / PLMN to the Internet are less common. There are reasons explaining this.
First, most Internet devices have no E.164 number. Second, the Law is not permitted in
most countries to route telephone calls over IP networks, because call monitoring and call
record checking mechanisms have not been well developed on IP networks. Another
problem is: the lack of interests from conventional telecom operators, since most IP
telephony is free of charge.
Using current technologies, such as ENUM (Mapping an E.164 number to URI using
DNS) [17] and SIP (Session Initiation Protocol) [16], a call can be initiated from GSM
networks and terminated to the Internet. But, ENUM is unsuitable to integrate the GSM and
IP networks for two reasons. First, ENUM using the DNS (Domain Name Service)
technology. That means ENUM servers maintain mapping records in cache that cannot be
updated in a real time fashion. As a result, an IP phone may not be reached when it is not
updated. Second, ENUM should be developed and maintained by an international
organization. This could take years to complete, and would be too late for the IP phone
market. For a telecommunication operator, we should design a solution to ease the using
all networks cannot be supported by ENUM. In this thesis, we present a solution for one
mobile phone number with parallel ring service in different networks. The solution consists
of four mechanisms: (1) SCP (Service Control Point) service logic for both GSM and IP
networks; (2) Location database for mobile SIP phones; (3) SS7 and IP networks call party
handling service logic (finite state machine design); (4) An added SS7 Q.763 parameter
carrying an IP phone’s URI between SS7 network nodes. With this solution, mobile phone
subscribers don’t need an extra IP phone number and can enjoy one personal number
service in both the GSM network and IP network.
Take Taiwan’s telecom environment for example, many people own more than one
number; one for the cellular phone (GSM), another for home use (PSTN). If the IP Phone
numbers have been assigned to identify the IP phones, another number added. If somebody
wants to reach a person who owns both the GSM cell phone and IP phone, he or she must
remember both numbers. It’s inconvenient. This thesis present a parallel ring service in both
networks (GSM and IP) to identify a user who owns GSM cell phone and SIP Phone.
Anybody who want to reach this person can only dial GSM MSISDN number and then both
phones (the GSM phone and IP phone) ring if both of them are online; any phone can
answers and the other one will be released. This kind of application save the communication
fee in the caller paid market, such as Taiwan. When a subscriber goes abroad and takes an
IP phone with him/her, he or she won’t paid the expensive international communication fee
and can do the same conversation with Caller. With the broadband internet environment
now, the IP phone can receive good quality for this kind of service.
Chapter 2 describe basic concepts of the GSM network routing and service architecture,
SIP, how these two networks interworking. Chapter 3 describes the detail GSM and IP
networks integration mechanisms and parallel ring service architecture system design. And
in Chapter 4, the implementation part is described.
1.1 The interworking between the PSTN and IP networks
using ENUM
Because of the law not permitted yet in Taiwan, the calls originated from PLMN/PSTN
and terminated to Internet Telephony Network are restricted. With the ENUM, this direction
call traffic can be done. Take Japan, US and Taiwan for example, they are waiting for the
Law’s open to this market (There still exist the police monitoring and call records issues and
they are considered should be solved before the IP Phone Numbers allotting). At first, they
require the identification number of the IP Phone been assigned to the operator or ISP
(internet service provider) if they want to use the ENUM technology. Second, who will
implement such an international ENUM DNS database is another problem.
Figure 1-1 describes how this model work:
3. The gateway routes the call to the SIP Proxy.
4. & 5. The Gateway converts the telephone number into an ENUM domain name and queries the ENUM.
6. The ENUM responds with the Internet address of the user.
7. The VoIP service provider will use the Internet address to complete the call to the VoIP phone.
Figure 1-1: PSTN-IP number mapping
1.2 The advantage of one-number parallel ring service
The wireless mobile phone users could use the one personal number service to be
reached easily. In spite of GSM or IP networks, user can own the both GSM mobile phone
and SIP mobile phone at the same time and chose to pick up the SIP phone to save money,
because the IP phone usage is free now. Especially for the International Roamer, they can
save lot of international fee while they go abroad.
One number to identify a person in a mobile world no matter what kind of phones
(GSM or IP) they use will meet the human’s habit. We believe this kind of service will be
accept by customers in very short time.
1.3 Why ENUM solution not suitable?
There’s some ENUM issues should be considered, these issue may cause the ENUM
unsuitable to this one number solution.
(1) ENUM use the DNS architecture, it is an overhead to the operator.
(2) There should be a independent and public permitted organization to develop and
maintain the ENUM database, it not mature yet. In this fast change world, it will be too late
to enter the IP phone market.
(3) DNS uses the Cache mechanism, it will cause the SIP URI update too late (not real
time). Especially lot of Internet Users use the dynamic IP Address ADSL in their activation
field.
For the operator, how to reduce the impact of the VoIP coming is an big issue. To
provide the service to IP phone with the customer’s original GSM MSISDN is one of the
solution to attract customer to stay in their network. But, since the ENUM has the issue not
solved yet, how operator could implement such a service? If they use this thesis’s solution,
the service can be provided immediately. Because (1) Operator can do the fast fresh SIP
location database to replace ENUM, and (2) Operator can handle and record the call traffic,
so billing and monitor (for Police investigation) is no problem ( to meet US government
VoIP consideration, they will make up the Law to restrict the VoIP phone numbers alloting).
1.4 Motivations, advantages and innovations of this thesis
The thesis focus on the GSM operator’s point of view to integrate the VoIP into the
PLMN/PSTN network nowadays. If is unnecessary to use an extra number to identify a
VoIP Phone, and ease the user behaviour of the GSM and Internet integrated telephony
networks. The idea is that everyone just has to use only one number to identify his or her
GSM / SIP Phone(s) and others can reach him/her with this only one personal identifier. The
four main mechanisms and it’s practice of this thesis will be detail described in chapters 4.
In the end of this chapter, the advantages and innovations of this thesis is also introduced.
The advantages of this thesis’s design include: (1) Reduces the Numbers use to identify
the Cell Phone and VoIP Phone. User can use only one number to identify both of the 2
different phones. Caller just have to remember only one number to dial and then can reach a
person who have both of the GSM cell phone and SIP WIFI Phone. User don’t have to
change their calling behavior while the dual network introduced to the telecommunication
world. (2) In Taiwan’s environment, this system design could reduced the impact to the
main GSM operators (Taiwan Cellular Corp., Chong-Hwa telecom, Fareastone Cellular
Corp.) while the VoIP domain becomes large. (3) Speed up the SIP URI Query time
(Operator self-owned DB should be faster, specially design using the share memory for
RAM Database and is 2-segments indexed)
The novel design of this thesis includes: (1) Paging the GSM Cell phone and VoIP
Phone at the same time (parallel ring service), it’s the pioneer service in this dual mode
network. (2) New designed (added) Q.763 ISUP Paramater for deliver Ineternet URI in the
SS7 network. (3) New designed SIP location database, and add Status field, its role like
VoIP’s HLR for the status and location query. (4) After the Query, SCP could get the GSM
cell phone and VoIP phone status at the same time. And then do the right paging strategy.
The IP phone status can carried with the new designed Q.763 parameter and the terminal
gateway could also get the same IP Phone status (Offline, Online, etc.) Thus the trunk
resource would be saved in operator owned network.