Single Radio Voice Call Continuity (SRVCC)

This paper compares three voice options for quality and efficiency:

1. Native VoLTE client integrated in the handset chip set. The number of commercial
handsets with native VoLTE clients is growing rapidly.

2. Non-native VoLTE clients. Third party applications which can register to IMS (IP
Multimedia Subsystem) and establish VoLTE call using QoS (QCI1). Examples include
Bria and CSipSimple.

3. Over-the-Top (OTT) VoIP applications such as Skype, Facetime and Viber.

The native VoLTE with Adaptive Multirate Wideband (AMR WB) provided an average
of 10.2 kbps with the codec rate of 23.85 kbps and 8.8 kbps with the codec rate of
12.65 kbps.

The SRVCC probability also depends on the parameter settings, like minimum
Reference Signal Received Power (RSRP), which define the threshold when the LTE
network initiates SRVCC

When the handset leaves the LTE coverage area, the VoLTE connection can be
handed over to a CS connection in a 3G or 2G network. This procedure is called
Single Radio Voice Call Continuity (SRVCC) or enhanced SRVCC (eSRVCC).

SRVCC functionality is available for QCI1 connections but not for OTT VoIP. If an OTT
VoIP connection loses LTE coverage, the call continues in the 3G or 2G network as a
VoIP connection over a best effort data connection, not as a CS call. VoIP in 3G can
provide reasonable voice quality in low loaded networks, but VoIP does not work in
practice on a 2G network. Also, the connection break during the inter-system
handover is substantially longer for best effort data than with SRVCC for QCI1 (less
that 300ms according 3GPP standard).

The AMR-WB data rate for CS connection ranges from 6.6 kbps to 12.65 kbps, while
the VoLTE connection can use data rates up to 23.85 kbps, enhancing the quality of
the connection compared to HD voice in CS networks.

The voice quality depends heavily on the voice codec sampling rate and the
resulting audio bandwidth. An AMR Narrowband (NB) codec provides audio
bandwidth of 80-3700 Hz, while an AMR Wideband (WB) extends the audio
bandwidth to 50-7000 Hz. Furthermore, handset acoustics may limit the maximum
bandwidth provided by the speech codecs. Terminal acoustic requirements can be
found in 3GPP TS 26.131.
The throughput measurements shown in Figure 3 include full IP headers. VoLTE
throughput requirements on the radio interface can be further reduced with Robust
Header Compression (ROHC,) which improves spectral efficiency. ROHC runs
between the base station and the handset. Figure 4 illustrates the benefit of RoHC -
the header size is reduced from 40 bytes to 5 bytes, which is relevant for voice
traffic since the voice packets are small. For example, an AMR 23.85 kbps voice
packet is 60 bytes and AMR 12.65 kbps just 32 bytes. The header can be larger than
the voice packet if header compression is not used. Therefore, activation of header
compression is essential for maximizing VoLTE capacity.
VoLTE uplink performance in the weak signal can be enhanced with TTI bundling
which allows the handset to repeat the same transmission in four consecutive 1 ms
TTIs. TTI bundling makes the uplink more robust and enhances coverage by 4 dB.
The benefit of TTI bundling in the weak signal is shown in Figure 5. The uplink Block
Error Rate (BLER) is reduced from 73% to 9% with TTI bundling. The low BLER
maintains good voice quality and avoids unnecessary retransmissions which eat up
substantial radio resources. TTI bundling is switched on only when the handset hits
the edge of the coverage area. TTI bundling runs between the base station and the
handset.

Note on IMS:
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem, IMS is an architectural framework for
delivering Internet Protocol, IP multimedia services. It enables a variety of services to be run seemlessly rather than
having several disparate applications operating concurrently.