I was not sure if I should post my “TFO and
TrFO introduction” here as there are probably only a handful of people on the
globe who are interested in this topic. Also, some prior knowledge is
required on how GSM and UMTS networks handle circuit switched calls and how
things change with the introduction of 3GPP Release 4 BICN functionality. A
good intro on this topic can for example be found in my book (take a look on
the left). Most of my blog entries require no such in-depth prior knowledge but
this one definitely does. I decided to do it anyway as I didn’t find any other
usable overview on the web.
Tandem Free Operation (TFO) and Transcoding
Free Operation (TrFO) Overview
Voice calls from mobile phones today always involve several voice transcoding steps
in GSM and UMTS Release 99 networks. This is done mainly to transport a voice
call, which uses sophisticated compression in the radio network, through the
core network with the standard G.711 PCM (Pulse Code Modulated) codec over 64
kbit/s circuit switched links. Furthermore, transcoding a voice data stream to the
64 kbit/s G.711 PCM codec is necessary in order to call fixed line phones.
Another advantage of voice transcoding in the network is to be able to process
the voice data stream and to compensate for unwanted effects such as echo in
the speech path.
With the introduction
of the Bearer Independent Core Network functionality in the 3GPP Release 4
standards, core network connections are based on high bandwidth ATM or IP links
rather than on 64 kbit/s TDM links. In such networks it is therefore possible
to transmit a voice data stream in the core network with other codecs than 64 kbit/s G.711. This
also has the additional benefit of removing transcoders from the speech path which
reduces speech path delay and improves speech quality to some degree as the
speech signal is no longer degraded by the transcoding process. Furthermore,
not having to convert to the narrow voice band G.711 codec allows the use of
AMR Wideband (AMR-WB) codecs. These codecs are not backwards compatible to
G.711 without loosing their much better audio quality which they achieve by
encoding twice the frequency range as previous Full Rate, Enhanced Full Rate,
AMR and G.711 codecs. A general downside of not using a transcoder in the
network is that it is no longer possible to process the voice data stream to
remove effects such as echo.
Transcoder Free Operation (TrFO)
Bearer Independent Core Networks (BICN), PCM links which require the use of the
G.711 codec are no longer used. Instead, connections to RNCs in the radio
network and connections to other Media Gateways are based on ATM and IP.
Therefore, the MSC servers involved in a call can establish a bearer path
without activating voice transcoders in the media gateways between two mobile phones. For a mobile to landline call the connection remains transcoder free up to
the media gateway which connects the wireless network to the fixed network.
Here, the transcoder is used to convert the compressed audio signal to G.711.
to mobile calls, both handsets report their codec capabilities to the MSC
servers involved in the call as part of the signaling before the bearer path is
established. Thus, an AMR-WB codec is automatically used if supported by both
terminals. This greatly improves the voice quality of the call. If not
supported on both ends, a standard narrow band AMR codec will be used for the
connection instead. This is called Transcoder Free Operation (TrFO) and the out
of band transcoder control performed by MSC servers is described in 3GPP TS
works well while both ends of the connection remain in the UMTS network, it has
to be deactivated if one of the two parties is handed over to a GSM network
during the connection. This happens, for example, if the subscriber roams out
of the UMTS coverage area or in case the UMTS coverage is temporarily weakened
by obstacles while GSM network coverage remains acceptable. This can happen for
a number of reasons like because of different locations of the UMTS and GSM
base station sites or due to better GSM in-house coverage. If AMR-WB was used for the call, deactivating TrFO
also requires a fallback to AMR which results in an abrupt voice quality
degradation of the call in case the GSM network does not support Tandem Free
Operation which is described next (see also 3GPP 23.153, chapter 6.5).
communicate with services in the networks such as a voice mail system, or a
prepaid top up voice server, DTMF (Dual Tone Multiple Frequency) tones are used
to send passwords and commands to the system. If a call is established from a
UMTS terminal, DTMF tones are sent as out of band signaling messages to the MSC
Server. The MSC Server is then responsible to forward the message. If the
service platform which analyzes the DTMF tones is BICN compatible, the tone
is delivered as a message. If the service platform is still using PCM links and
the G.711 codec, the DTMF tone is inserted into the voice connection at the Media
Gateway at border to the PCM network. If a call is established from an external
PCM network to a service in BICN network, DTMF tones are not sent as messages
but inside the voice bearer. In this case, 3GPP 23.153, 5.9 requires the
gateway at the border of the BICN network to discover the inband DTMF tone,
remove it from the voice stream and send a DTMF signaling message to its MSC
Server which then forwards the DTMF message out of the voice band to the
Tandem Free Operation (TFO)
networks, TrFO can not be used even if the core network uses BICN media
gateways and MSC-servers. This is because the A-Interface, which connects the media
gateway to the BSC/transcoding unit is based on PCM. Thus, voice calls from GSM
radio networks always arrive at the media gateway as a PCM G.711 64 kbit/s
circuit switched stream. Unlike in UMTS, where the transcoding is a
functionality of the media gateway, GSM uses a dedicated transcoding unit
between the BSC and the media gateway. This is necessary in order not to change
the radio network architecture when the core network is upgraded to BICN. As a
consequence, a mobile to mobile GSM call will always have two PCM links in the
connection, one to and from each transcoder and thus a GSM call is always
established using the G.711 codec. The two transcoders of the connection are
also called a tandem. In order to remove the transcoder tandem from such a connection,
in band codec negotiation between the two transcoders can be attempted once the
call is established. Instead of sending signaling messages via the MSC-servers
through the network, the two transcoder units send their information embedded
in the G.711 link. This is done by using a special bit pattern as part of the
voice stream which is recognized by the other end as a transcoding control message
and not as a part of the voice signal. If both transcoder units support at
least a single common codec like AMR 12.2 or an AMR-WB codec, the 64 kbit/s
G.711 connection is used to tunnel the compressed and encoded voice stream. As
the audio stream is compressed most of the bits of the transparent 64 kbit/s
stream are not used. This effectively removes the transcoder tandem from the
voice connection and is thus called Tandem Free Operation (see 3GPP TS 23.053
and 3GPP TS 28.062). While audio quality and delay times can be improved this
way, no bandwidth savings can be achieved in the core network unlike with TrFO.
in the network can temporarily interrupt TFO and the speech path in the core
network automatically falls back to standard G.711 encoding. This is the case
for handovers for example in which the call is handed over to a cell which is
controlled by a different TRAU. This is the case for an inter-BSC handover for
example. Once the handover is complete, TFO has to be negotiated again between
the two TRAUs (see 3GPP TS 23.053, 6.5.7). TFO is also temporarily interrupted
when the MSC inserts DTMF tones or announcements. As the MSC is not aware of
TFO it overwrites the compressed speech information in the circuit connection.
This is detected by the terminating transcoder and transcoding elements are
automatically reinserted into the speech path. Especially for AMR-WB, TFO has
to be started again as soon as possible in order to avoid speech quality
bridges are another network feature which automatically disable TFO. This is
because standard conference bridges mix G.711 voice signals from several
parties to produce a combined signal which also overwrites the TFO data stream.
This again forces the TRAUs to introduce a transcoder in the speech path again.
While this has little impact on narrow band speech calls from a voice quality
perspective, it is not possible to use AMR-WB when a conference bridge is
inserted unless the conference bridge also supports TFO and the AMR-WB codec.
There we go, if you have made it up to here, please consider leaving a comment 🙂
16 thoughts on “Deep Inside the Network: TFO and TrFO”
Greetings; we are just about to introduce MSC servers and Media Gateways in our network and your artilce (blog) has helped answer quite a few grey areas. Thanks. Live long and prosper.
Hi Martin, this was a very useful read about TFO and TrFO. Please feel free to post other documents that expand on how the media gateways are changing as we move away from circuit switched networks.
This is a very succinct yet informative write-up on TFO/TrFO. Thanks!
Between this and your other article on AMR-WB, there is some ambiguity about whether calls that use AMR-WB are amenable to TFO operation. Can you shed some light on this?
For AMR-WB you need TrFO if the call runs over UMTS and TFO if the call runs over GSM.
if an operator want to define AMR codec on BSC ,,
and want to know the percentage of subscribers devices supporting this.. what is the distinguishing in handset ???
the BSC, or rather the MSC, can see if a mobile is capable of AMR or not by analyzing the Vocoder Preference List which should be part of the Bearer Capability Information Element for example in a SETUP message. The mobile is AMR capable if the list includes Full Rate Codec 3 and Half Rate Codec 3.
Good explanation of a tricky subject, thanks Martin. You might consider what happens when a mobile changes AMR codecs mid-call, e.g. does it renegotiate end-to-end ?
In TrFO call how what is the measure of Voice Qulaity on BSC as BSC is only shipping frames it recieves from mobiles to MG on packet network. Are there standard practices to measure Voice Qulaity on TrFO call.
I haven’t been involved in Voice Quality testing before. One of the companies I know that is doing this is Polystar for example. In case you are interested, go to their website and check them out.
Excellent review, thanks.
Is TFrO and TFO mainly a NW funcationality? As far as UE end, if UE can support AMR rate change dynamically then it can work with TFrO/TFO enabled NW?
thanks for the comment. TFrO and TFO are network functionalities. But what they do is to allow AMR capable mobile to communicate directly instead of with transcoders in the network. That means that if a mobile wants to change the codec it will not tell the network but directly the mobile on the other side.
There are still some questions open for me on this topic like for example the following: In GSM, the network can decide to put a connection on a half rate air interface channel. This means not all codecs will work anymore. So I wonder how the mobile on the far side knows that it can no longer use certain codecs because the other mobile is on a half rate channel. So if anybody knows the answer to this question, please leave a note.
Very good Review. Thanks
Hi Martin. Thanks for this explanation. I have a simple question. Enabling TrFo in a network, could impact terminals? How does a terminal behaves? Thanks and regards.
in theory it should be transparent to mobiles. However, I wonder how mobile B knows that mobile A is only on a half rate channel and can thus not support the higher AMR codecs. Haven’t had the time to dig that deep.
Thanks, very good and helpfull posts (especially about TrFO/TFO, which most of the normal people do not read 🙂 ) .
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