Page 365 - From GMS to LTE

VoLTE, VoWifi and Mission Critical Communication 351

Enhanced Voice Services (EVS) codec, which offers another significant speech‐quality
improvement. Among VoLTE‐capable devices, at least AMR‐WB, or, if supported by the
two devices and the network, EVS, is typically chosen, as these offer a much better sound
quality than the traditional narrowband codec. Many circuit‐switched 3G and even
some 2G networks and mobile devices also support AMR‐WB today so it can be used
when establishing a connection to legacy circuit‐switched networks and devices as well.
Some network operators also support a wideband speech codec in their fixed‐line IP‐based
networks so AMR‐WB is the codec of choice for such connections, too. In this scenario,
however, a media gateway is required at the border between the networks as the wideband
codec used in fixed‐line networks (G.722) is different and more bandwidth intensive (64
kbit/s) than AMR‐WB (G.722.2) used in mobile networks, which typically uses 12.65 kbit/s.
If AMR‐WB is not supported by one party, AMR‐NB is chosen as the lowest common
denominator. From a bandwidth point of view there is almost no difference as AMR‐NB
also requires a datarate of 12.2 kbit/s.

Adaptive Codecs
In addition, codecs are rate adaptive today. AMR‐WB, for example, can encode a
voice data stream at a rate between 6.6 and 23.85 kbit/s. At the lower end, sound
quality is rather limited while encoding a speech signal sampled at 16.000 Hz at 23.85
kbit/s gives a superb result. In practice, most network operators choose to limit
AMR‐WB to 12.65 kbit/s as it seems there is little gained in terms of speech quality
beyond that datarate. Another reason for using 12.65 kbit/s is that 2G and 3G cir-
cuit‐switched networks also limit AMR‐WB to this datarate as it fits into the original
AMR‐NB channels.
The idea behind datarates lower than 12.65 kbit/s is that if network coverage gets
weak, a slower datarate increases the robustness of the transmission and more
speech packets might make it to the other side in time. Which datarate is used at
any particular time is decided by the speech coder in the mobile device. The codec
rate can be changed every 20 ms and each speech frame encapsulated in an IP
packet contains a header that informs the receiver which datarate is used. Figure 5.9
shows how the datarate is signaled in an IP/UDP/RTP (Real‐time Transport
Protocol) speech packet.

Codec Selection
While each device in a call has the freedom to change the codec’s datarate whenever
required, the type of codec is negotiated only once during call setup. This is done in two
steps. In the first step the originator of the call includes information about supported
codecs in the SIP Invite message. At the end of this message all supported codecs are
listed in the Session Description Protocol (SDP) part. For details see RFC 4566 [11].
Here is an abbreviated example:

m=audio 42888 RTP/AVP 116 118
a=rtpmap:116 AMR-WB/16000/1
a=fmtp:116 mode-change-capability=2;max-red=0
a=rtpmap:118 AMR/8000/1
a=fmtp:118 mode-change-capability=2;max-red=0