Paolo Cennamo, Antonio Fresa, Anton Luca Robustelli, Francesco Toro
Co.Ri.TeL, Via Ponte Don Melillo, I-84084 Fisciano (SA), Italy
Maurizio Longo, Fabio Postiglione
Dipartimento di Ingegneria dell’Informazione ed Ingegneria Elettrica
a degli Studi di Salerno, Via Ponte Don Melillo, I-84084 Fisciano (SA), Italy
Beyond-3G networks, IP Multimedia Subsystem, Secure Real-Time Protocol, multimedia communications,
voice quality.
The emerging all-IP mobile network infrastructures based on 3rd Generation IP Multimedia Subsystem philos-
ophy are characterised by radio access technology independence and ubiquitous connectivity for mobile users.
Currently, great focus is being devoted to security issues since most of the security threats presently affecting
the public Internet domain, and the upcoming ones as well, are going to be suffered by mobile users in the
years to come. While a great deal of research activity, together with standardisation efforts and experimen-
tations, is carried out on mechanisms for signalling protection, very few integrated frameworks for real-time
multimedia data protection have been proposed in a context of IP Multimedia Subsystem, and even fewer
experimental results based on testbeds are available. In this paper, after a general overview of the security
issues arising in an advanced IP Multimedia Subsystem scenario, a comprehensive infrastructure for real-time
multimedia data protection, based on the adoption of the Secure Real-Time Protocol, is proposed; then, the
development of a testbed incorporating such functionalities, including mechanisms for key management and
cryptographic context transfer, and allowing the setup of Secure Real-Time Protocol sessions is presented;
finally, experimental results are provided together with quantitative assessments and comparisons of system
performances for audio sessions with and without the adoption of the Secure Real-Time Protocol framework.
The very rapid evolution of the communication infras-
tructures has progressively rendered access to com-
munication facilities ubiquitous.
These fast-evolving communication technologies
have greatly stimulated research activity on security
issues, encompassing both data confidentiality and
data protection, both in corporate and residential en-
While, on the one hand, the legacy mobile digital
networks (GSM, GPRS, UMTS) provide strong secu-
rity and confidentiality guarantees, on the other, the
emergence of the 3rd Generation IP Multimedia Sub-
system (IMS) as the unified and standard platform,
based on the all-IP paradigm for the provision of real-
time multimedia services both to mobile and fixed
users, is bringing the security issues to the forefront
once again. Indeed, the adoption for next-generation
mobile networks of an IP-based transport infrastruc-
ture, based on Internet Engineering Task Force (IETF)
protocols, both for signalling, based on Session Initia-
tion Protocol (SIP) (Rosenberg et al., 2002), and mul-
timedia real-time data transport, by using Real-time
Transport Protocol (RTP) (Schulzrinne et al., 2003),
will expose future mobile telecommunication infras-
tructures to all the security threats (and maybe new
ones) of the public Internet. This emerging scenario
requires specific research and definitions of solutions
aiming to guarantee acceptable levels of user data
confidentiality and protection.
Another key feature of future IMS-based networks
will be the access-domain independence, i.e. the IMS
service provision infrastructure will be totally inde-
pendent of the particular radio technologies deployed
in the access network. In other words, future IMS
will be access-agnostic and it will work on the top
of any kind of wired or wireless access technology;
Cennamo P., Fresa A., Luca Robustelli A., Toro F., Longo M. and Postiglione F. (2007).
In Proceedings of the Second International Conference on Security and Cryptography, pages 125-132
DOI: 10.5220/0002118601250132
on the other hand, given that each access technology
provides different security guarantees (varying from
very strong to none at all), the IMS cannot in general
rely on such capabilities. Then, IMS-specific security
mechanisms must be provided, and such mechanisms
can only operate from the IP Layer upwards (network,
transport or application), being IP the first common
technological layer envisaged by the IMS philosophy.
The paper is organized as follows. First of all, we
point out the advantages of adopting appropriate se-
curity protocols in order to protect both the signalling
and the real-time multimedia flows, for which we fo-
cus on the Secure RTP protocol (SRTP) (Baugher
et al., 2004); then, we propose a architectural solu-
tion to involve security mechanisms during sessions
establishment and control and we describe the de-
veloped testbed which implements it within an IMS-
like prototype. Finally, we provide some quantita-
tive evaluations and comparative assessments related
to voice quality parameters measured in end-to-end
audio sessions, pointing out the the influence of the
SRTP framework deployment on voice communica-
tion quality.
Most researchers consider IMS as the key element in
the next generation network architectures since it en-
ables the convergence of data, speech, and mobile net-
work technologies over a unified IP-based infrastruc-
ture. The organization responsible for the definition
of Beyond-3G (also known as B3G) mobile commu-
nication systems, including IMS, is the Third Gen-
eration Partnership Project (3GPP) (3GPP, The 3rd
Generation Partnership Project, 1998). The 3GPP has
chosen SIP as the signalling protocol for the setup,
modification and tear-down of multimedia sessions.
The Call Session Control Function (CSCF) servers
represent the core elements, within the IMS, for the
management of the SIP signalling. The Proxy CSCF
(P-CSCF), usually located in the Visited Network,
represents the first contact point for the user termi-
nal towards the IMS network and takes care of for-
warding the SIP signalling towards the subscriber’s
Home Network; the Serving CSCF (S-CSCF) is prob-
ably the main CSCF server and is located in the sub-
scriber’s Home Network (typically the operator to
which the user is subscribed): its task is to process
the SIP signalling, take decisions on managing the
multimedia sessions. Another important function of
the IMS architecture is the Home Subscriber Server
(HSS) database that contains all the user-related sub-
scription data required to handle a multimedia ses-
sion, such as information on user location, security
data and user profiles. The interaction among the
three CSCF nodes and the HSS allows the complete
management of the SIP signalling necessary for the
establishment and support of the multimedia sessions.
Nowadays, millions of customers are using com-
puter networks for e-banking, e-commerce and sub-
mitting their tax returns and since 3G architecture
aims to enable such secure transactions together with
real-time services in its IP-based infrastructure, the
security issues have acquired a primary importance.
Network security problems can be roughly divided
into six closely related areas, each of them with its
peculiar goals:
Authentication: to guarantee user identity;
Confidentiality: to keep information out of the
hands of unauthorised users;
Integrity: to avoid information alteration or the
whole substitution of messages by malicious
Non-repudiation: to avoid that users deny having
sent or received information actually sent or re-
ceived by them;
Authorization: to allow only authorised users to
access particular resources and services;
Availability: to guarantee the effectiveness of a
service avoiding actions of disturbance by mali-
cious users.
There are many possible approaches in order to
provide security services; indeed, security features
can be implemented in different layers of the TCP/IP
reference stack: at the Network Layer by adopting
IPsec (Thayer et al., 1998), at the Transport Layer
by TLS (Dierks and Allen, 1999) and at the Appli-
cation Layer using HTTP Digest (Franks et al., 1999)
or other.
Security in an IMS scenario can be categorized as
follows (Koien, 2002):
Access security: it includes mutual authentication,
encryption and integrity of both signalling and
multimedia data which are exchanged between
the B3G terminal and the network;
Network security: it deals with traffic protection
between network nodes, which can belong to the
same operator or different ones.
The IMS adopts IPsec for signalling protection
both in the access and network domains but nothing
is specified for data or multimedia traffic. In order
to accommodate all requirements (very different and
SECRYPT 2007 - International Conference on Security and Cryptography
Figure 1: A SRTP message.
sometimes very stringent) of data and real-time appli-
cations, security protocols adapted to the single ap-
plication (and thus working at the Application Layer)
appear more appealing.
In particular, four factors must be taken into ac-
count in order to protect multimedia real-time com-
munications: bandwidth availability, delay, com-
putational power of the mobile terminals and
transmission-error sensitivity. To address these issues
a very promising choice is the adoption of the Secure
Real-time Transport Protocol (SRTP) (Baugher et al.,
2004). This protocol employs particular transforms
such as the Advanced Encryption Standard (AES)
symmetric cipher (Schaad and Housley, 2002), since
symmetric cryptography is characterised by lower de-
lays and computational burden with respect to asym-
metric cryptography (Stallings, 2004); furthermore,
such a ciphering system can avoid the error propaga-
tion drawback if used in a stream modality. Another
advantage of SRTP is related to bandwidth consump-
tion due to the frequent re-keying procedures occur-
ring during long real-time sessions. In fact, it intro-
duces a 32-bit RollOver Counter (ROC) in order to
expand the space of the RTP sequence numbers and
eliminate the need of re-keying (Blom et al., 2002).
Alternative proposals are available for multime-
dia real-time protection based on IP tunneling proto-
cols, such as IPsec, which seem to suffer some perfor-
mance limitations (Ranganathan and Kilmartin, 2001;
Vaidya et al., 2005).
The SRTP protocol was designed to be deployed in
heterogeneous network architectures; the critical fac-
tors which were taken into consideration were band-
width, delay, the need of computational resources and
transmission errors. SRTP is an RTP profile and it
can be considered as a sub-layer implementation lo-
cated between the RTP application protocol and the
transport protocol: on the sending side, SRTP first
intercepts RTP packets and then forwards equivalent
SRTP packets; on the receiving side, it first intercepts
SRTP packets and then relays equivalent RTP packets
upwards. On the other hand, the Real-time Transport
Control Protocol (RTCP) is secured by Secure RTCP
(SRTCP) so as SRTP does to RTP. Message authenti-
cation based on SRTCP is mandatory when SRTP is
used; moreover, it can protect the RTCP fields to keep
track of session members, it can provide feedbacks to
RTP senders and securely manage counters of packet
Then, SRTP provides a framework for authentica-
tion and encryption of RTP and RTCP data streams.
Specifically, SRTP proposes a set of default crypto-
graphic algorithms and it also allows for the introduc-
tion of new ones in the future. Together with appro-
priate mechanisms for key management, SRTP can
effectively provide security services to RTP applica-
tions both of unicast and multicast transmissions.
Fig. 1 depicts the format of a SRTP message. The
specific additional fields introduced by SRTP are:
Master Key Identifier (MKI): the key management
mechanism defines and uses this field. MKI iden-
tifies the master key from which the session keys
can then be derived. Authentication and/or en-
cryption of the RTP packets is then performed by
using such session keys.
Authentication Tag: this field is employed in order
to carry message authentication data. The Authen-
ticated Portion of an SRTP packet is made up of
the RTP header followed by the Encrypted Por-
tion of the SRTP packet. If both encryption and
authentication are applied, encryption must be ap-
plied before authentication on the sending side
and vice-versa on the receiving side. The Authen-
tication Tag provides authentication of the RTP
header and payload, and it also provides, even if
indirectly, replay protection by authenticating the
sequence number. It is worth noting that the MKI
is not integrity-protected since this would provide
no additional protection.
3.1 The Cryptographic Context
Each SRTP stream requires the sender and the re-
ceiver to maintain cryptographic state information;
moreover, in order to establish an SRTP session two
Figure 2: The proposed IMS-based architecture for secure
multimedia communications.
users have to come to an agreement on specific pa-
rameters, such as the cryptographic and integrity tech-
niques to use, the master key from which the session
keys will be derived and so on. All this information
is called Cryptographic Context and is handled by a
key management mechanism external to SRTP. Sev-
eral key management standards have been proposed
for SRTP cryptographic contexts, such as MIKEY
(Arkko et al., 2004) and KEYMGT (Arkko et al.,
In this Section we present the architecture of a
Beyond-3G network built by integrating the SRTP
framework into the IMS infrastructure. The adoption
of SRTP to protect real-time communications is mo-
tivated by performance limitations in IPsec-based so-
lutions, as already stated in Sect. 2. Furthermore,
SRTP allows to propose a novel technique for the key
exchange mechanism, based on IMS SIP signalling,
which does not require any additional message thus
increasing the overall performance (see Sect. 4.2).
4.1 The IMS-based Testbed
As previously sketched, with the present work we aim
to show how it is possible to integrate a secure archi-
tecture for multimedia communications into a B3G
network scenario. In order to introduce the several
actors involved in such a scenario, it might prove use-
ful to briefly illustrate the environment we adopted for
the developed testbed.
Our IMS-like prototype is represented in Fig. 2,
where two mobile phones are included which are
connected to the IMS Home Network (i.e. with the
S-CSCF server) through the P-CSCF nodes of each
mobile user’s Visited Network. The Authentication,
Authorisation, Accounting (AAA) server (Senatore
et al., 2004) is introduced according to the IMS ar-
chitecture defined by 3GPP. In the depicted network
scenario, when a user switches his mobile phone on,
a registration phase takes place by means of the AKA
protocol (AKA, 2003) encapsulated within SIP REG-
ISTER messages. This protocol allows a mutual au-
thentication (that is the user and the network authenti-
cate each other): through this procedure each user in-
dependently computes the cryptographic and integrity
keys which will be used in subsequent secure commu-
nications. Furthermore, during the registration phase
an IPsec security association is established between
each user and its reference P-CSCF in order to guar-
antee a strong protection for the subsequent SIP sig-
nalling messages.
4.2 A IMS-based Master Key Exchange
RFC 3711 provides two different methods for select-
ing the Master Key that can be used during an SRTP
session: the first mechanism proposes the use of the
MKI of the SRTP packet header, while the second
provides the definition of a (From, To) mechanism,
as already explained in Subsect. 3.
In our IMS testbed, we adopt a novel mechanism
for cryptographic context transfer which does not in-
troduce either a new protocol or a new messages ex-
change, which would be both a burden to the sig-
nalling system and a cause of delay; in fact, all of the
information necessary to establish the SRTP session
can be encapsulated within the SIP signalling mes-
sages, in appropriate fields of the Session description
Protocol (SDP) (Handley and Jacobson, 1998) body,
already conveyed by SIP messages for multimedia
In particular, we integrated into our B3G network
prototype a mechanism to transfer the cryptographic
context based on the encapsulation of context infor-
mation into the SIP INVITE transaction. Fig. 3
shows the whole signalling process taking place be-
tween two end users for a multimedia session set-up.
An important role is played by the S-CSCF and
the AAA server; indeed, by parsing the SIP INVITE
body the S-CSCF detects the presence of the crypto-
graphic context attribute in the SIP message and con-
sequently sends a specific request to the AAA server,
SECRYPT 2007 - International Conference on Security and Cryptography
INVITE (crypt context)
INVITE (crypt context)
INVITE (crypt context)
INVITE (crypt context)
200 OK (crypt context)
200 OK (crypt context)
200 OK
(crypt context, K, T )
200 OK
(crypt context, K, T )
ACK ( K , T )
ACK ( K , T )
ACK ( K , T )
ACK ( K , T )
Figure 3: A SIP-based Master Key exchange mechanism.
which answers back by a message with the session
key and its lifetime for the under way SRTP session.
Then, the S-CSCF includes these parameters into the
200 OK message and subsequent ACK message con-
cluding the INVITE transaction. At this point, the two
users are able to establish an SRTP session. When the
timer expires, a re-INVITE message is sent between
the users in order to re-negotiate the cryptographic
context and the previously described procedure takes
place once again.
4.3 A SRTP-enabled Voice Application
Besides the IMS-based solution, in this paper we pro-
pose and develop a Voice over IP (VoIP) application
that implements the SRTP framework by modifying
the open-source Robust Audio Tool (RAT) code, ver-
sion 4.2.25 (Robust Audio Tool (RAT), 2004). In the
application we propose for secure real-time commu-
nications, it is possible to distinguish two different
the SIP module for the signalling and session con-
the RAT module for the audio communication
These two modules need to exchange information
using a particular communication channel: in our so-
lution, such modules are organized according to the
schema reported in Fig. 4.
The SIP module receives the Master Key within
the SIP signalling flow, as described in Sect. 4.2. By
means of a local Message Bus (Mbus), such a key is
then transferred to the RAT Controller which forwards
it to the Media Engine: thus, this Master Key becomes
the Active Key for the actual SRTP session.
Let us recall that the SRTP layer is located, within
the protocol stack, between the Transport Layer (in
this case, UDP) and the Application Layer (RTP).
It is worth pointing out that, during the definition
phase, we evaluated two different approaches to the
integration of SRTP in the VoIP application: the for-
mer aimed to maintain RTP and SRTP as separate
modules, the latter to create a hybrid RTP/SRTP struc-
ture. At the end, we decided upon the latter solu-
tion since it requires less computational burden and
achieves better performances, which are crucial con-
straints for an application that has to process real-time
4.3.1 The SRTP Transmission Phase
Our implementation does not modify the RTP pack-
ing phase and operates on the packet which is ready
to be sent. The packet is “intercepted” inside the
rtp send data()
function and, if the session encryp-
tion is enabled, the SRTP packet setting phase starts.
The SRTP packet setting phases implemented in
our prototype are those provided by RFC 3711. The
first phase consists of the search of the active key
by using the
find key()
method: this phase is
strictly related to the exchange key mechanism de-
scribed in Sect. 4.2 and uses the key exchanged dur-
ing the SIP session setup. The second phase con-
cerns the generation of the session keys by means
of the
key derivation()
method through the ac-
SIP Modue
SIP Modue
Figure 4: A schematic representation of the developed
Voice over IP application.
tive key obtained in the first stage. The third phase
schedules the keystream generation by invoking a
keystream generator()
function. In the fourth
phase, the payload is encoded by performing a sim-
ple XOR operation between the generated keystream
and the payload itself. The resulting encrypted pay-
load replaces the non-encrypted RTP packet payload,
since their length are the same. The last phase sched-
ules the Authentication Tag generation by means of
hmac SHA1()
function. This tag is appended to
the RTP packet so that the receiver can authenticate
the packet itself by it.
When those phases are accomplished, the Con-
troller switches back to the
rtp send data()
tion, that delivers the packet thus prepared to the
udp send()
function, which finally sends the UDP
segment toward the receiver.
4.3.2 The SRTP Reception Phase
Similarly to what happens at the beginning of the
transmission phase, the received packet is intercepted
rtp receive data()
function and the SRTP
session management starts.
The first action to be performed is finding the ac-
tive key by using the
find key()
method, as de-
scribed for the transmission phase. This key is
then passed to the
key derivation()
method in or-
der to generate the session key. At this stage the
hmac SHA1()
function locally generates the Authen-
tication Tag to be compared with the one received
within the SRTP packet. If the two tags match, the
received packet is authentic and it is thus possible to
go on with the decryption process. The next step is
devoted to the estimation of the ROC related to the Se-
quence Number of the RTP packet. This process takes
place after the packet authentication procedure and
it is needed in order to estimate the correct counter
Isolated Fast Ethernet LAN
VoIP communication
Figure 5: The voice quality measurement testbed.
value of the SRTP packet.
keystream generator()
generates the keystream that is used in the
dencript payload()
method to execute de-
cryption. Subsequently, the control switches back
to the
rtp receive data()
method and the audio
content can be reproduced by the application as
In line of principle, the introduction of the SRTP
framework introduces, both at the transmission and
reception side, an increase of complexity and compu-
tational load that might potentially badly influence the
system performances. That is why it appears particu-
larly worthwhile to analyse the impacts of our SRTP
implementation on a real end-to-end audio communi-
cation, as reported in the following section.
In order to assess the influence of our SRTP imple-
mentation on real VoIP communications, we compare
quantitatively system performances in terms of both
the mean audio delay and the mean perceived voice
quality at the receiver.
The measurement testbed is shown in Fig. 5,
where the VoIP applications involved in the audio
communications are connected to the same (isolated)
Fast Ethernet LAN segment (no signalling server is
involved). In such an environment, the delay in-
troduced by the network can be considered negligi-
ble. Measurements are collected by an Agilent Voice
Quality Tester (VQT) connected, by means of proper
Phone Adapters, to the audio card line-in of PC
to the audio card line-out of PC
in the same system
SECRYPT 2007 - International Conference on Security and Cryptography
0 1000 2000 3000 4000 5000 6000
time (sec)
One-way delay (msec)
) (SRTP, GSM codec)
) (no SRTP, GSM codec)
Figure 6: Typical one-way delay time series provided by
VQT for application running on PCs not specialized for
real-time applications. The adopted codec is GSM.
configuration during audio sessions both in case VoIP
applications implement the SRTP framework, as de-
scribed in Sect. 4, and in case of no SRTP implemen-
tation. The VQT can represent measurements results
both in a textual (log files) and a graphical way.
and PC
, where VoIP applications are run-
ning, are both Linux-based machine with a Pentium
IV 2.8 GHz processor and audio cards, whose quality
can heavily influence quality measurements, are both
Creative SoundBlaster Live!.
We select two audio codec available in the VoIP
applications under test: (full-rate) GSM and G.711
implementing the companding µ-law algorithm (Bel-
lamy, 2000). Every transmitted packet contains 20
msec of speech, i.e. the RTP payload is 33 byte or
160 byte long for GSM and G.711, respectively. The
encryption algorithm used by SRTP in our measure-
ments is AES in Counter Mode (AES-CTR).
A key parameter that influences the quality of
real-time communication is the one-way delay expe-
rienced by two speakers during an audio session, also
known as mouth-to-ear delay (Jiang et al., 2003). In
order to assess the impact of SRTP on the total delay,
we collect N = 1000 measures of audio delay for each
session (one using SRTP and one without SRTP for
each audio codec), where every delay sample Θ(n) for
n = 1, ..., N is computed every = 6 sec by the VQT
by estimating the position of the maximum of the
cross-correlation between the transmitted signal and
the received one. The measured mouth-to-ear delay
time series for GSM and G.711 µ-law are indicated
as Θ
(n) and Θ
(n), respectively, while the presence
of SRTP is pointed out by the superscript (e). Typ-
ical delay time series (with and without SRTP sub-
layer) are shown in Figs. 6 and 7 for VoIP applications
0 1000 2000 3000 4000 5000 6000
time (sec)
One-way delay (msec)
) (SRTP, G.711
-law codec)
) (no SRTP, G.711
-law codec)
Figure 7: Typical one-way delay time series provided by
VQT for application running on PCs not specialized for
real-time applications. The adopted codec is G.711 µ-law.
running on operating systems not specialized for real-
time applications, such as Linux or Windows, where
general purpose schedulers can cause some artifacts,
such as a piecewise linear decrease (or increase) of
the one-way delay.
The computed average delays, reported in Table
1, seem to indicate that AES-CTR encryption has no
significant influence (just few milliseconds).
Another key parameter to assess audio communi-
cation quality is the perceived quality of the speech
at destination. A widely adopted tool for objec-
tive measurements of it is the Perceptual Evaluation
of Speech Quality (PESQ), described in ITU-T Rec.
P.862 (Beerends et al., 2002), which uses a sensory
model to compare the transmitted signal with the re-
ceiving one. In order to relate its results to the tra-
ditional subjective quality score Mean Opinion Score
(MOS), based on time-consuming human listeners in-
terviews, the PESQ Listening Quality (PESQ-LQ) is
often used, providing values ranging from 1 (bad) to
4.5 (very good).
The average PESQ-LQ values were computed on
30 voice clarity measurements, provided again by the
VQT using English speech samples, and are reported
in Table 2, where it is possible to notice that the SRTP
does not introduce any appreciable variation on the
quality of the speech. This agrees with the general
conception that encryption in itself should not cause
Table 1: Average mouth-to-ear delays (in msec).
GSM G.711 µ-law
no SRTP 132.71 144.14
138.86 149.79
Table 2: Average PESQ-LQ.
GSM G.711 µ-law
no SRTP 3.693 4.028
3.690 4.013
an information loss.
Summing up, the introduction of the SRTP frame-
work does not seem to influence speech quality from
a practical prospective in our prototype testbed. Then,
SRTP seems suitable for a deployment in real-world
network scenario.
The introduction of real-time cryptography technique
for the multimedia flows with the adoption of the
SRTP protocol is aimed to guarantee a good security
level to multimedia communications.
One of the mechanism that offer a good level of
robustness against the two time pad typologies of at-
tacks is the introduction of a periodic key update
mechanism. In our proposal the update mechanism
towards IMS SIP signalling does not introduce any
increase of the number of exchanged messages, as
it may happen adopting a Master Key Identifier or a
(From,To) mechanism.
The quality of the communication does not turn
out to be degraded even though a real-time cryptogra-
phy and de-cryptography is performed. In particular,
the SRTP framework does not influence the quality
of the speech during VoIP communications, both in
terms of delay and PESQ-LQ index.
Future developments will concern, first of all, the
practical establishment of an SRTP session also for
the video content between two users within the IMS-
like prototype. Another development will be related
to the adaptation of our architectural solution to a
multi-conferencing scenario.
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Baugher, M. et al. (2004). The secure real-time
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