Carrier Aggregation
LTE Categories
Voice over LTE
Session Initiation Protocol (SIP)
IP Multimedia Subsystem

New Functions for LTE

LTE still did not meet the requirements of IMT Advanced. In particular, the required high data rates could not be achieved.

• 100 Mbit/s for moving participants
• 1 Gbit/s for fixed stations

The theoretical maximum bit rates of LTE were 100 Mbit/s and 150 Mbit/s (Cat 3 and Cat 4). This applied to 2×2 MIMO operation. There was also Cat 5 with 300 Mbit/s, but it only worked with 4×4 MIMO, which was only applicable in stationary operation with LTE routers. In order to further improve the transmission rate, it was necessary to allow higher bandwidths. This was achieved by combining different carriers with each other. This principle is called Carrier Aggregation (CA).

Carrier Aggregation

Typically an operator (of LTE) has more than one carrier available. An operator may have two carriers available in one band and can combine them. These can vary in bandwidth. E.g. 20 MHz and 10 MHz. If these bands are directly next to each other, this is called intra-band contiguous aggregation. If the bands do not lie together, this is called non-contiguous aggregation. The carriers can also be in different bands. New bands have been added with LTE. In Europe, bands 7 (2.6 GHz) and band 20 with 800 MHz.

Carrier aggregation was introduced with the so-called Release 10 for LTE. This specification was started back in 2010 when LTE had not yet been introduced in most regions. This release was available in 2011. At the end of 2013 there were first tests with Vodafone and LTE Advanced in Dresden, with band 20 being bundled with band 7. Starting in 2014, LTE Advanced was slowly rolled out in metropolitan areas.

Theoretically, with LTE Advanced, up to 5 carriers can be combined to provide a bandwidth of 100 MHz. However, until today this was more of a theory. In practice, two bands are brought together. At Deutsche Telekom, for example, two bands in band 3. At Vodafone, a 20 MHz carrier in band 7 and another carrier in band 20.

With LTE Advanced, the requirements of IMT Advanced were met. In stationary operation, a data rate of 3 Gbit/s is possible with 5 carriers and 8 x 8 MIMO. However, this is more of a theory. However, it is realistic that 1 Gbit/s is possible under favorable conditions.

For end devices, CA initially has the obvious advantage that (theoretically) higher data rates can be achieved. There is another, arguably even more important advantage for the eNodeB. There is more flexibility in the allocation of resource blocks, especially if many participants are active in a cell. This means that practically all participants can be distributed across different carriers without having to instruct the end devices to switch to a different carrier. In this way, the capacity of a cell can be better used to the benefit of everyone.

LTE Categories

A transition from LTE to LTE Advanced was possible for most eNodeBs through SW updates. This is more difficult with user equipment. Even with simple LTE, there were different performance classes that an end device could fulfill. She was marked with different “Cat”s. These are shown for Release 8 and Release 10 in the following table.

LTE Categorie	Rel	Bitrate Down Mbit/s	Bitrate Up Mbit/s	Maximum Number Carrier	QAM	MIMO
Cat 1	8	10	5	1	16/16	1×1
Cat 2	8	50	25	1	64/16	1×1
Cat 3	8	100	50	1	64/16	2×2
Cat 4	8	150	50	1	64/16	2×2
Cat 5	8	300	75	1	64/64	4×4
Cat 6	10	300	50	2	64/16	2×2, 4×4
Cat 7	10	300	100	2	64/16	2×2; 4×4
Cat 8	10	3000	1500	5	256/64	8×8

Typically, stationary UE such as LTE routers had higher categories because only they were able to enable around 4×4 or even 8×8 MIMO. Normal handhelds and smartphones were initially only able to achieve 2×2 MIMO and therefore had lower bit rates. It wasn’t until the end of the noughties that the first smartphones with 4×4 technology came onto the market.

Voice over LTE

LTE was a pure IP based network. There was no access to the public telephone network as was still the case with GSM and UMTS. When making normal voice calls, GSM or, if necessary, UMTS was still preferred. However, the UMTS network was increasingly being switched off because it was not as efficient as LTE and people wanted to use the corresponding bands for LTE (refarming). GSM should also be scaled back more and more. Here too, voice transmissions are actually very inefficient meanwhile.

Therefore it was required to route voice services over LTE or IP. These services already exist on the Internet. It is called Voice over IP (VoIP). Two things must be guaranteed here. First: the identity of the user and its „authorization“. In addition, a voice connection requires a guaranteed data rate and low latency. After all, a call cannot be interrupted if there is a lot of traffic. If response times become too long, a fluent conversation is no longer possible. The second thing is called Quality of Service. This allows certain services on the Internet to be prioritized.

SESSION INITIATION PROTOCOL (SIP)

There are various options for Voice over IP, some public (puplic domain), others such as SKYPE proprietary. The best-known protocol is the Session Initiation Protocol (SIP). The central element here is a SIP registrar, a server with a subscriber database. A SIP end device, e.g. a PC with special software, the so-called user agent (UA), connects to the registrar and registers there. The UA is authenticated and registered by storing its IP address and SIP ID in the database. If this is the case, the UA can either start a conversation with another participant or be called himself.

Depending on whether a single IP service or several IP services are used, the following conversations go through one or two proxy servers. It is also possible to reach the landline telephone network via gateways. The actual transmission of speech takes place using the Real Time Transport Protocol (RTP). This ensures smooth voice transmission with low latency and a guaranteed data rate. This happens by determining the quality of service in the network.

IP MULTIMEDIA SUBSYSTEM

The IP Multimedia Subsystem has been continuously specified by 3GPP since the early 2000s. It is based on the SIP and has the following goals:

• Handling general packet-switched connections
• Connections from IP-based networks to line-based networks such as classic telephone networks (PSTNs)
• Support for various media types (voice, video,…)
• Provision of sufficient bandwidth and short latency (Quality of Service)
• Service-dependent cost determination

The service-dependent cost determination and the provision of the quality of service (QoS) take place in the EPC via the so-called Policy and Charging Rule Function (PCRF). This also checks which services a UE is allowed to use according to its contract. Voice over LTE is an additional service.