IMS functionality and Kamailio in a nutshell
The complexity upon the integration of VoLTE into the LTE-core network relies upon an old acquaintance, the IP Multimedia Subsystem (IMS).
IMS is a standard architecture of the mobile telephony which is bound to the SIP (Session Initiation Protocol) and has been elaborated on since many years. Until now, integrate providers, such as Vodafone, O2 or Telekom, have primarily implemented the IMS standard. They are offering either fixed-line- or mobile telephony, and are working on providing the solution for all the networks. In Europe, the IMS has until lately been deployed in the fixed-line networks, in the mobile and radio communication and the “usual” technology, the transmission was particularly grid bound.
IMS is developed on the basis of the standardizing panel 3GPP (3rd generation Partnership Project) to the network-comprehensive transmission service in All-IP-Networks. Due to its general orientation, the IMS is a broadly defined standard, which may be interpreted differently and employed from one vendor to another. In practice, this causes the incompatibilities to appear, which normally require some configuration adjustments. Integration of the Ericsson-components proved to be rather seamless, whereas the Nokia networks are turning isolated, when IMS core of any other network is being used. The removal of such incompatibilities is merely subject to analysis costs.
Individual components of the network are mainly featuring the SIP-standard: users have to go through the authentication and localization stages, and register in the SIP-system, then start working from the next available proxy. There is only one big difference that signals are standardized and the telephony-server is also exchanging with the spare network infrastructures, so that, for instance, the prioritizing is established. Vendors, such as Ericsson, Huawei or Nokia are providing comprehensive solutions for VoLTE, which might, however, be rather expensive.
Besides the Diameter-Protocol employed in IMS, in most cases, the SIP-standard is also using the Radius-Protocol, which adds up to numerous functions. While the radius-protocol is used merely for authorization, authentication and accounting (AAA), the diameter-protocol is deployed in the real-time accounting (Ro/Online Charging), in the status retrieval (for example, registration), or saving or retrieving application-specific information. Furthermore, the diameter-protocol is used in the information exchange during the ongoing conversation.
In the meanwhile, the telephony servers remove the relevant information out of the signals, such as IP-addresses, ports, codecs and the medium (Audio or Audio/Video) used. This information obtains the Packet-Core (PCRF) on the standardized diameter-interface (Rx). The PCRF can now perform the connection and, in turn, refer the necessary QoS-parameters to the radio cell. As we deal with the bidirectional protocol (the diameter-protocol is bidirectional), the radio cell signals to the telephony-server with the help of diameter, when the device loses the radio (communication contact), without signing off.
The IMS has already established itself as the mobile radio standard, thus, it is used by vendors as a large common denominator in the LTE-network architecture. In theory, it is absolutely possible to install the products of various vendors or employ the license-free open-source solutions, such as Kamailio as VoIP-core. Yet, their installation will definitely be subject to some configurations, which are varying from one vendor to another.
Kamailio is a leading open-source platform aimed at performing various services on the SIP-basis. Taking the role of a classic telephone switch, Kamailio is the open-source software, which is protected by the GNU Public License (v2). In addition, some parts of the code are also protected by the BSD-License.
The purpose of the Kamailio project lies in developing of a solid and scalable platform. It can be used in the Vo-IP periphery and the SIP-Proxy as well, registration service or an application server. The scope of its applicability ranges from the embedded systems, such as DSL-router in the telephone system up to large installations with multi-million users in the services of internet providers – Kamailio covers the whole telephony spectrum.
The telephony connection is locally held and is optimally spread all over the network. Conversations within Hamburg are not led through the network access point in Munich, their processing takes place at the access point, located in Hamburg. This directs the conversations to the particular Gateway in other networks, which might also be based in Hamburg. In order to provide a high quality control, the end-devices are generally providing the RTCP-Data, which can be monitored.
For the reliable data security, voice transmission and network-management operate from the Internet through separate APNs (Access Point Name) and, furthermore, can be encrypted with the IPsec. The APN can be compared to the separate DSL-connection, which is used exclusively for the voice transmission (Source: Wikipedia). Therefore, the accruing volumes of data upon the voice connection can be billed independently from the regular data volumes used by customers.
The access login data of the terminal device to the SIP-telephone switch cannot therefore be manipulated: it is encrypted and stored in the SIM-card. With this data authorization by the telephone switch into the end-devices for the indicated conversation takes place and can, therefore, be accessed by using a mobile number.
However, the network access point is not stored in the SIM-card, and it is normally assigned upon the connection establishment (similar to the assignment of the NTP-server upon DHCP). The network access point corresponds to the radio cell, and, as a rule, it is geographically closely located to the user.
Current challenges, characterizing the implementation of the standard, lie in the backlog demands, which are inherent to the compatibility of vendors’ products. Non-homogeneous infrastructures are rather sophisticated to use, because many vendors are embedding their own “specialties” into the product, apart from the standard ones.
In addition, the standard’s full practical operation is being hindered with another piece of complexity. For example, when dealing with the service quality, one has to provide a telephone number, even though the SIM-card ID might serve for the same purpose. The phone number is being matched with the SIM-card, which requests the information from the profile database. This creates an unnecessary technical request, and consequently, a conditional burden.
Source: Translated from „Voice over LTE. VoIP in Mobile radio network: theory, technology and practice“ by Wolfgang Drost0