Initial Development of a SIP-/RTP

16. VDE/ITG Fachtagung Mobilkommunikation
18. - 19. May 2011 - Osnabrück, Germany
(2011-04-21)
Initial Development of a SIP-/RTP-based Core Network
for the TETRA Mobile Radio System aiming at Transparent
Availability of its Features in LTE
Daniel Hartmann, Mark Stephan, Xing Cao, Diederich Wermser (Research Group IP-based Communication Systems, Ostfalia University of Applied Sciences, Salzdahlumer Str. 46/48, D-38302 Wolfenbüttel).
Michael Zeuschner, Roberto Hunger, Filipp Andjelo (Rohde & Schwarz Professional Mobile Radio GmbH,
Fritz-Hahne-Str. 7, D-31848 Bad Münder).
Abstract
This initial paper introduces basic questions and alternative approaches to replace today’s circuit switched TETRA core network with an NGN-based core network.
The paper presents first results from an R&D project, which is carried out in collaboration of Rohde & Schwarz
Professional Mobile Radio GmbH, the Research Group for IP-based Communication Systems at Ostfalia University of Applied Sciences in Wolfenbüttel and the Institute for Communications Technology at TU Braunschweig, Germany. The project is funded by the Federal Ministry of Economics and Technology (BMWi, Germany) in context of the ZIM program (Zentrales Innovationsprogramm Mittelstand).
1
Introduction
Packet switched networks, especially those based on
the Internet Protocol, are constantly advancing and
spreading in and to various application areas like IPTV or public mobile radio (e.g. LTE). Along with that,
it gets easier and more flexible to integrate specific
service and performance features into these IP-based
networks or to interconnect one network with another.
Due to these considerable advantages, there are approaches to replace the circuit switched TETRA core
network by an NGN-orientated packet switched core,
using typical VoIP-protocols like SIP and RTP. TETRA is the digital, professional mobile radio system
that is commonly used by public authorities like Police or Fire Brigades, public transport organisations or
industrial companies [1].
Section 2 gives a brief overview of the current TETRA system, while section 3 focuses on the approach
for an NGN-based core network, covering basic questions concerning mobility support for TETRA devices, PMR-specific services (in particular push-to-talk
group calls) and integration with LTE.
2
Today‘s TETRA System
Compared to public mobile radio, TETRA offers a
different spectrum of services. Although being capable of normal full duplex calls as in telephony, the
main focus of TETRA is push-to-talk involving
groups of users. Moreover, unlike mobile phones in
GSM or UMTS, TETRA mobile stations have the
ability to establish ad-hoc point-to-point connections
without the need of a base station [1].
The original core network as well as the air interface
of TETRA is based on circuit switched components
using TDM (Figure 1). This came along with the
digitalization of communication systems (e.g. ISDN)
that used interfaces with 2048 kbps linespeed or multiple of these [1].
The air interface for TETRA, typically working in the
range of 400 MHz, is strictly standardised, while the
interfaces between elements of the core network are
not standardised [1]. Therefore, vendors implemented
their own proprietary solutions for the core network.
Multiple BSs (also referred to as Location Areas, LA)
can be merged to a cluster, typically managed by a
cluster controller (CC). Ideally this task can be taken
over by any BS inside the cluster to provide redundancy in case of a drop out.
Figure 1: Today’s circuit switched TETRA
3
Several aspects of the new VoIP-based core have to be
discussed, if it is to take over all functionality of the
former TETRA core. The most important ones are the
mobility support for the mobile devices (section 3.2)
and TETRA-specific services, in particular group call
concepts (section 3.3). According to experience,
group calls typically make up to about 90% of traffic
in a TETRA installation.
Approach for an NGN-based
Core Network
Implementing a VoIP-based and NGN-orientated core
network for TETRA results in exchanging a circuit
switched network with a packet switched network,
commonly based on the Internet Protocol. Established
VoIP protocols like SIP and RTP will be used for signalling and user data, respectively.
Using an IP-based network infrastructure offers more
flexibility to deploy or extend a TETRA core network,
since IP networks are widely spread. Furthermore,
VPNs, MPLS and QoS mechanisms allow several
logical networks to co-exist on the same physical
network and to adapt each one of them to its specific
requirements, which is critical for a voice orientated
system like TETRA.
Speaking of “service creation”, the advantage of
NGNs is to make additional IP-based features and
services available using application servers, as described in the IMS concept.
3.1
Architecture
All components and protocols associated with the
TETRA air interface will be left unchanged. Therefore, the access network will stay circuit switched and
gateways are needed in order to connect it to the new
packet switched core network. These gateways will be
located inside the base stations (BS) as a logical unit
(Figure 2). The radio connection terminates on one
side of the gateway, while the VoIP connection terminates on the other side.
Note that this scenario is different from the IMS concept, where the IP connection terminates in the MS
and mobility is provided by tunnelling via an additional IP layer. Section 3.2 will give a more detailed
description of this topic.
Figure 2: TETRA-to-VoIP BS with gateways
Preferably, the core network should consist of standard SIP network elements. In terms of the IMS, all
TETRA specific features that are not supported by
these standard components will be implemented with
additional application servers, for example a group
call or paging server.
3.2
Mobilty Aspects
3.2.1
Comparison with IMS concepts
Mobility in the IMS is implemented by using an additional IP layer (Mobile IP [10]) to encapsulate the
payload and tunnel the traffic to the correct location
area (Figure 3). While roaming, the applications on a
mobile device always use the same IP address and behave like a static node.
With the new TETRA core network, the IP connection
terminates in the gateway located in the BS and not in
the MS. Therefore, the Mobile IP concept cannot be
applied, since the SIP UA on the VoIP side of the
gateway is not mobile itself. Because of this, mobility
has to be implemented on a different layer, e.g. the
SIP layer. Possible solutions could be SIP mechanisms
like REFER, Re-INVITEs or the Presence Service
(NOTIFY, etc.).
Figure 5: Handover w/ and w/o new cluster
Figure 3: Mobility in UMTS Release 5 [9]
In GSM or UMTS the handover decision is done by
the network and the mobile devices merely provide
the related measurements (Figure 4). In TETRA a
handover is initiated by the mobile devices themselves [1].
Even more timing critical is scenario (3) (Figure 6),
which shows two devices changing their BS at the
same time, while being in a session with each other.
This is not unlikely, since a group of Firemen for example will pass radio cell borders almost simultaneously, while being on a fire run.
Figure 6: Simultaneous handover (same cluster)
3.2.3
Conceptual Sequence Diagrams
A concept for a handover (1) exemplified by the REFER method is shown in Figure 7 (some details left
out for better overview).
Figure 4: Handover decision in GSM [8]
3.2.2
Handover Scenarios
Due to the hierarchical structure, several handover
scenarios have to be considered. Additionally, there
has to be a differentiation between handovers while or
while not being in an existing session.
In TETRA, an MS is not obliged to inform its old BS
1 in case of a handover [1]. Therefore, after associating with a new BS 2, it has to be made sure to notify
the old BS 1 and the cluster controller of the handover. After that, the call can be transferred to BS 2 with
identical session parameters, initiated by BS 1.
While being idle, a handover can basically be reduced
to a simple location update in the appropriate location
register (e.g. SIP Registrar). Handovers within existing sessions are much more complex and timing critical, since no user data is to be lost. In the course of
proof-of-concept implementations, it is to be examined, if SIP-based mobility is fast enough to provide
seamless handovers.
In the basic handover scenario (1) (Figure 5) an MS
moves to a different BS connected to the same cluster.
Moving to a different cluster is shown in scenario (2),
as well.
Figure 7: Concept of a handover using REFER
Figure 8: Conceptual group call scenario
3.3
Group Call Concepts
3.3.3
TBCP and SIP RE-INVITE
The group call is a prominent service in a TETRA
system. The following section illustrates the group
call as a service example, implemented by additional
application servers.
A TETRA group call session can stay alive for a much
longer period of time than a typical VoIP call, with
changing sources and destinations of simplex media
streams within the same session.
3.3.1
One current focus of examination is to analyse,
whether to control the RTP streams in a group call
VoIP session via the Talk Burst Control Protocol
(TBCP) or via SIP RE-INVITE messages. These are
two alternative approaches to implement TETRA
group call functionality.
Terminology
First of all, a clarification of the term “group call” is
needed. In a classical telephony environment, a group
call causes a group of TEs to ring at the same time. As
soon as one user accepts the call, the other TEs stop
ringing and a normal two-legged call is in progress.
Contrary to this, a group call in TETRA means paging
a live voice message to a group of participants simultaneously, without users accepting the call manually.
This is a service much similar to PoC functionality
(Push-to-Talk over Cellular) defined for the IMS [7].
3.3.2
TBCP is based on RTCP (Figure 9) and is used to
handle access control for all participants of a session.
TBCP consists of typical messages like Request,
Granted or Deny [7].
TETRA group call
The TETRA group call is initiated by a push-to-talk
(PTT) button on the MS. If no prior group call is existent, this is triggering a request directed to the network to allow the user to speak. Once the permission
is granted, a simplex point-to-multipoint media stream
is established. Note that releasing the PTT button does
not end the session. The session is kept alive, until the
initiator ends it manually. Until then, every member of
the call group is able to cast a message in the way as
described above.
To implement automatic answering of a UA without
the need of manual user interaction, SIP provides an
additional header field Answer-Mode, as described in
RFC 5373.
Figure 9: TBCP stack [7]
With RE-INVITES on the other hand, it is possible to
control RTP streams in an existing session. This is
done by the SDP attributes “sendrecv”, “sendonly”,
“recvonly” and “inactive”, as described in RFC 4566.
3.3.4
Group Call / Paging Server
In a VoIP scenario using a SIP proxy, it is not intended
to control routing of an RTP stream by a separate instance. This is typically done by the underlying IP
layer and an RTP stream is typically exchanged directly between the participating TEs.
In order to implement a paging functionality, additional components (paging and group call server) are
needed to multiply an incoming RTP stream and to
dispose it to the appropriate destinations (Figure 8).
The group call functionality preferably has to be realised with standard VoIP mechanisms in combination
with application servers.
3.5
Integration with LTE
“LTE for TETRA” can be used to extend its range to
areas not covered by the TETRA radio network, as
well as to extend its features and services. Moreover,
LTE provides much more bandwidth than the TETRA
air interface. This enables additional services, e.g.
transferring high resolution pictures to a Police task
force spontaneously.
In parallel to the definition of LTE as the evolution of
the radio access network, SAE (System Architecture
Evolution) specifies the corresponding evolution of
the core network [5]. The design of interworking between the new NGN-like TETRA core network and
the LTE/SAE network needs thorough analysis to ensure best possible support for TETRA-specific service
features.
Figure 10: SipXecs architecture [2]
3.4
Proof-of-Concept Implementation
4
Conclusions
The proof-of-concept implementation of the new
TETRA core network as explained above will be
based on sipXecs components as far as possible.
This initial paper introduces basic questions and aspects of designing an NGN-orientated core network
for the circuit switched TETRA system.
SipXecs is an open source unified communications
platform developed by SIPfoundry [17]. It has a modular architecture and makes use of standard interfaces
using SIP, XML or SQL, thus implementing the idea
of a SIP Service Orientated Architecture (SSOA).
Signalling and media related components are separated, analogue to IMS concepts. Connections to nonVoIP architectures are provided by external gateways
(Figure 10).
Gateways are needed to connect the circuit switched
TETRA air interface to the packet switched core network. The gateways can be located in the base stations and consist of a TETRA UA and a SIP UA.
Due to the modular architecture and its interfaces,
sipXecs is very scalable and is capable of distributed
operation with high availability mechanisms.
Mobility for TETRA radio devices cannot be implemented by using Mobile IP concepts like in the IMS,
since the IP connection terminates in the BS and not
in the MS. Instead, mobility can be implemented using SIP-based mobility support.
TETRA-specific services like group calls are to be
implemented with additional application servers.
5
Outlook
Along with first proof-of-concept implementations,
subjects like signalling delays, scalability and stress
tests will be focused more intensely. Especially delays
to initiate a TETRA group call with SIP signalling are
a critical topic, since a group can comprise several
hundred participants.
Furthermore, it is important to be able to record and
store user data to verify spoken statements at a later
point. This is basically feasable, since application
servers like the paging services have access to the respective RTP streams anyway.
Speaking of LTE, so far it is planned to use it as an
additional and independent access network aside the
TETRA air interface. The next step-up could be the
implementation of a seamless handover for mobile
stations between both access networks. For this,
SRVCC-like concepts [14] may be considered, which
handle handover procedures from circuit switched to
packet switched access networks.
6
References
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Digital Mobile Communications and the TETRA
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[4] Poikselkä, Mikka et al.: The IMS. West Sussex:
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[5] Dahlman, Erik et al.: 3G Evolution. 2nd Edition,
Burlington: Academic Press, 2008.
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3G IP Multimedia Subsystem (IMS). Second
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Reprint November 2006.
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packet Systems (EPS). West Sussex: John Wiley
& Sons Ltd, 2008.
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[9] Trick, Ulrich; Weber, Frank: SIP, TCP/IP und
Telekommunikationsnetze. 4. Auflage, München: Oldenbourg Verlag, 2009.
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[12] Olsson, Magnus et al.: SAE and the Evolved
Packet Core. Burlington: Academic Press, 2009.
[13] Ericsson White Paper: Voice over LTE.
http://www.ericsson.com/res/docs/whitepapers/v
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[14] Salkintzis, Apostolis et al.: Voice Call Handover
Mechanisms in Next-Generation 3GPP Systems.
IEEE Communications Magazine, Feb. 2009.
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[17] SIPfoundry
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7
ACD
B2BUA
BS
CDR
CSV
FDM
HA
IMS
IP
ISDN
LTE
MPLS
MS
NGN
PMR
PoC
PTT
QoS
RPC
RTCP
RTP
SAE
SIP
SOAP
SRVCC
SSI
SSOA
TBCP
TDM
TE
TETRA
UA
VPN
VoIP
XML
Abbreviations
Automatic Call Distribution
Back-to-Back User Agent
Base Station
Call Detail Records
Comma-separated Values
Frequency Division Multiplex
High Availability
IP Multimedia Subsystem
Internet Protocol
Integrated Services Digital Network
Long Term Evolution
Multiple Protocol Label Switching
Mobile Station
Next Generation Network
Professional Mobile Radio
Push-to-Talk over Cellular
Push-to-Talk
Quality of Service
Remote Procedure Call
Realtime Transport Control Protocol
Realtime Transport Protocol
System Architecture Evolution
Session Initiation Protocol
Simple Object Access Protocol
Single Radio Voice Call Continuity
Short Subscriber Identity
SIP Service Orientated Architecture
Talk Burst Control Protocol
Time Division Multiplex
Terminal Equipment
Terrestrial Trunked Radio
User Agent
Virtual Private Network
Voice over IP
Extensible Markup Language