FR3037461A1 - METHOD FOR EVALUATING THE QUALITY OF MOBILE IP VOICE COMMUNICATION - Google Patents

Abstract

The present invention relates to a method for evaluating the quality of a Voice over IP communication in an access network. Said method is remarkable in that it comprises the following steps implemented by a mobile terminal: - taking into account positive and negative acknowledgments relating to packets transmitted or received by the mobile terminal within a virtual channel dedicated to voice, and - determine a rate of non-reception of said packets. Application to the switch from Voice over LTE to Voice over WiFi.

Description

The present invention relates to conversational services (voice communications and videophone communications), which are operated by means of a mobile terminal in a network to be connected to a mobile network. packet switching. More particularly, the present invention relates to the switching of such a communication from a first network to a second network when the quality of the communication is too bad.

In packet-switched networks (PSs), the same terminal, when multi-mode, can be connected and registered to the Internet Protocol (IP) core network through several accesses, such as a GSM network (Global System for Mobile Communications), a UMTS (Universal Mobile Telecommunications System) network, a set of DSL lines (Digital Subscriber Line), an Ethernet network (ISO / IEC 8802-3 standard). ), a WiFi network (IEEE 802.11 standard), or an LTE network (Long Term Evolution). A multimode terminal offers the user the possibility to choose a network among the different access networks compatible with the terminal to establish a communication. The choice criteria applied are generally the responsibility of the user or operator of the IP core network: the operator's price policy, the quality of the communication, the bandwidth available, and so on. A multimode terminal also enables its user to switch during a call from an access network (so-called initial) to another access network (called target), if for example the quality of transmission on the network of access. initial access becomes mediocre. This is known as a switchover of the communication ("handover" in English) between the initial access network and the target access network, or mobility PS-3037461 2 PS. This switchover may concern all or part of the media streams associated with the communication and / or signaling: this is called total or partial transfer of the communication. The mechanism of switching between two types of radio access networks (known as "Radio Access Technologies" or RATs) standardized by the 3GPP (Third-Generation Partnership Project) is controlled by the networks. It exploits radio measurements made in access networks, including signal strength, such as the RSRP (initials of the words "Reference Signal Received Power" meaning "Power of Reception of a Reference Signal") in LTE, and the quality of the signal, such as RSRQ (initials of the words "Reference Signal Received Quality" meaning "Quality of Receipt of a Reference Signal") in LTE. This mechanism is as follows: the mobile terminal measures the radio conditions of the downlink and reports them to the base station; the base station measures the radio conditions of the uplink link; English), and - the base station, based on the radio conditions and the policies provided by the operator, decides whether to switch the communication to another access network. The Evolved Packet Core (EPC) core architecture was designed to be compatible with both traditional 3GPP-standard access network types and non-mobile access network types. 3GPP (for example, WiFi type). Compatibility with non-3GPP accesses is described in 3GPP TS 23.402. Mobility can be seamless or stitched, on all connections or only on a few specific connections. The "Access Network Discovery and Selection Function" (or ANDSF) has been introduced in the "Release 8" version of 3GPP 3037461 3 standards to assist operators. to define policies concerning the choice of the network on which to route a given traffic. In "Release 8", the ANDSF defines access network information, in particular for improving the way in which a mobile terminal discovers new non-3GPP access networks. ANDSF also provides mobility policies to enable a mobile terminal to choose the most appropriate radio technology at any time. According to "Release 8", when a device discovers the presence of a non-3GPP network, either the traffic remains entirely on the 3GPP network, or the traffic is completely switched to this non-3GPP network, regardless of the type of traffic considered; however, since release 11 of the 3GPP standards, it is possible to switch a subset of traffic (regardless of the IP address and destination port) with specific characteristics (for example, voice communication).

Today, there is a great interest in LTE (Voice over LTE) and WiFi (Voice over WiFi) telephony services. videotelephony services, in particular on LTE ("Video over LTE", or ViLTE, in English). It has therefore been envisaged to allow the switching of a telephone / videophone conversation from one of these types of networks to another, with the aim of offering a high voice quality in both directions of communication. Furthermore, it is preferable to keep the mobile terminals on the LTE network as much as possible, because it is possible to guarantee the Quality of Service (QoS) (latency, jitter, and so on). on a LTE network, but not on a WiFi network. Unlike intra-3GPP mobility, switching decisions from a 3GPP access network to a non-3GPP network (eg from LTE to WiFi) are not taken by the network, but are taken by the mobile terminal. without help from the network. In particular, in the case of a telephone or videophone communication, the measurement of the radio conditions on the downlink in the access network used is conventionally performed by the mobile terminal on the basis of a RSRP (mentioned above). above) ; however, this signal is transmitted at different levels according to the topology of the network and the operators, so that the measurement of the level of this signal does not by itself make it possible to evaluate the radio conditions reliably. As regards the radio conditions on the uplink, the mobile terminal conventionally has no knowledge of it, whereas this uplink is generally the critical link for the quality of the communication, since the data rate is lower than the bit rate on the link. downlink. The present invention therefore relates, according to a first aspect, to a method for evaluating the quality of a Voice over IP communication in an access network. Said method is remarkable in that it comprises the following steps implemented by a mobile terminal: - take into account positive and negative acknowledgments relating to packets transmitted or received by the mobile terminal within a dedicated virtual channel to the voice, and - determine a rate of non-reception of said packets.

Thus, the present invention proposes, in order to evaluate the voice quality of a communication, to take advantage of the following characteristics of certain mobile networks: the existence of a mechanism for acknowledging data packets in the link layer of data ", that is, layer 2 of the Open Systems Interconnection (OSI) model, and the existence of a virtual channel dedicated to the voice. Indeed, as explained in detail below, there is a close correlation between the voice quality of the communication, estimated for example by the MOS (initials of the words "Mean Opinion Score" 30 meaning "Average Opinion Score") , and the rate of non-reception of 3037461 5 voice packets, estimated for example by the BLER (initials of the words "Block Error Rate" meaning "Rate of Error on Blocks", which measures the rate of non-reception radio frames). Thanks to these arrangements, the acknowledgments related to other "data" services than telephony / video telephony will have no effect on the evaluation according to the invention. Moreover, it is possible to calculate a rate of non-reception of radio packets for both the up link and the downlink. It will also be noted that the method according to the invention is advantageously independent of the services executed in parallel on the mobile terminal, as well as network policies implemented by the operators. According to particular features, said method further comprises a step of determining, based on said non-reception rate and the type of codec used by said mobile terminal, a speech quality level of the communication. Thanks to these provisions, the invention provides a precise and objective evaluation of the quality of a voice communication. According to a second aspect, the invention relates to a method for assisting the decision to switch between access networks. Said method is remarkable in that it comprises the following steps implemented by a mobile terminal: - determining, for a telephone or videophone communication on an access network, a rate of non-reception of packets as briefly described ci above or, respectively, a voice quality level as briefly described above, and - comparing said packet non-reception rate with a packet non-reception threshold depending on the type of codec used by the mobile terminal, or respectively, said voice quality level with a voice quality threshold.

Thanks to these arrangements, a multimode mobile terminal is helped to decide whether or not to switch a telephone or videophone communication to another access network, by enabling this mobile terminal to determine whether the voice quality on the network initial access is sufficient or not. It should be noted, however, that the switchover decision may be based on other factors besides voice quality, such as a minimum level of a reference signal on the target network. It should be noted in this respect that communications in a target WiFi network can, even when they benefit from a very good signal strength, suffer from a bad bit rate due to congestion of the link to the core network ( Backhaul), or being blocked by a firewall, so that switching to this WiFi network from an LTE network would in practice cause a loss of QoS. It is therefore desirable that the mobile terminal can still have other decision factors, such as the result of a rate or QoS test on the target network (for example by triggering a traffic surge) conducted in parallel with communication on the initial network. Correlatively, the invention relates to a mobile terminal. Said mobile terminal is remarkable in that it comprises means for: - taking into account, when said mobile terminal is engaged in a Voice over IP communication on an access network, positive and negative acknowledgments relating to packets transmitted or received by the mobile terminal within a virtual channel dedicated to the voice, and 25 - determining a rate of non-reception of said packets. According to particular features, said mobile terminal further comprises means for determining, on the basis of said non-reception rate and the type of codec used by said mobile terminal, a voice quality level of the communication.

According to other particular features, said mobile terminal comprises means, as briefly described above, for determining a packet non-reception rate or, respectively, means, as briefly described above, for determine a voice quality level when said mobile terminal is engaged on a telephone or videophone communication over an access network. This mobile terminal is notable in that it further comprises means for comparing said packet non-reception rate with a packet non-reception threshold depending on the type of codec used by the mobile terminal, or, respectively, said voice quality level with a voice quality threshold. The advantages offered by these mobile terminals are essentially the same as those offered by the correlative methods briefly described above.

Note that it is possible to make these mobile terminals in the context of software instructions and / or in the context of electronic circuits. The invention also relates to a computer program downloadable from a communication network and / or stored on a computer readable medium and / or executable by a microprocessor. This computer program is notable in that it includes instructions for executing the steps of the communication quality evaluation method or decision support method succinctly set forth above, when are run on a computer. The advantages offered by this computer program are essentially the same as those offered by said methods. Other aspects and advantages of the invention will appear on reading the detailed description below of particular embodiments given by way of non-limiting examples. The description refers to the figures which accompany it, in which: FIG. 1 graphically represents the BLER and the MOS as a function of the propagation loss in the uplink, for two different codecs; FIG. 2 schematically represents the virtual channels of an LTE network; FIG. 3 represents an embodiment of the method for evaluating the voice quality of a communication according to the invention, and FIG. 4 represents an embodiment of the method of tipping decision aid according to the invention. The present invention proposes to calculate, within a mobile phone, the rate of non-reception of packets intended to be transmitted by a mobile terminal to the core network (uplink), or by a network core 15 to a network. mobile terminal (downlink), within a virtual channel of packets containing speech frames. We can then (if necessary) easily deduce a voice quality level of communication. Indeed, as can be seen in Figure 1, there is, given the type of codec used by the mobile terminal, a close correlation between voice quality and BLER. In Figure 1, the two bottom curves represent the BLER as a function of the propagation loss in the uplink (expressed in dB), for two different AMR WB (Adaptive Multi-Rate-WideBand) codecs; the two upper curves represent, for the same two codecs, the MOS as a function of the propagation loss in the uplink (it is recalled that the MOS is a standardized, semi-subjective measure of voice quality). More precisely, this figure represents the "residual BLER", for the evaluation of which the number of radio frames to be transmitted is compared with the number of radio frames actually received; in other words, for the purposes of this evaluation, it does not matter, when a packet has been received, to know how many identical packets have to be transmitted to achieve this (it should be noted that, in a manner known per se, the heart of network informs the mobile terminal, when establishing the connection, the maximum number of repetitions that can be used during this connection). The "residual BLER" should not be confused with the "BLER MAC" (defined in 3GPP TS 34.121), for the evaluation of which the number of radio frames actually transmitted is compared with the number of radio frames actually received. It has been found experimentally that "residual BLER" is more closely correlated with MOS than "BLER MAC". In any case, it is evident from the curves in Figure 1 that the decline in MOS is closely correlated with the increase in BLER. The invention is generally applicable to any Voice over IP (VolP) protocol for distinguishing packets containing voice frames from data packets (collectively called "data") associated with other services. As an example of a non-limiting embodiment, the application of the invention to the VoLTE and ViLTE protocols will be described below, after a few reminders on these protocols.

As explained by the Wikipedia on-line encyclopedia, VoLTE and ViLTE are based on 3GPP LTE and LTE Advanced standards in the "Release 10" and "Release 11" versions; the VoLTE has been defined in the "PRD IR.92" standard, and the ViLTE in the "PRD IR.94" standard, by a group of equipment manufacturers and manufacturers of 25 telecommunication terminals forming the GSMA consortium (GSM Association) ; VoLTE and ViLTE use the "IP Multimedia Subsystem" (IMS) architecture and the "Session Initiation Protocol" (SIP) signaling protocol for call origination and voice transport in the LTE core network. . The setting up of 30 calls (numbering) and the voice transport on the LTE network use specific IMS profiles which guarantee a higher priority (better quality of service) for voice streams. This architecture allows the transport of voice communications (control plan and media plan) and SMS as IP streams similar to other data streams (eg, Internet, photos or videos) transported over LTE networks. VoLTE and ViLTE thus benefit from the high performance of LTE, which offers three to five times more bandwidth (voice and data) than UMTS networks; in addition, VoLTE and ViLTE 10 allow, compared to other VoIP techniques, to preserve radio bandwidth and decrease latency. To ensure compatibility with other mobile networks and older phones, the VoLTE standard requires mobile devices to be compatible with the AMR-NB 15 (Adaptive Multi-Rate-NarrowBand) narrowband codec already in use in GSM networks. UMTS; however, the voice codec recommended for VoLTE is the AMR-WB broadband codec (mentioned above), also known as "HD Voice" and defined by the ITU G722.2 standard. The AMRWB codec, when used end-to-end, allows a voice bandwidth that ranges from 50 Hz to 7 kHz. 3GPP has also defined in Release 12 a full-HD codec for a bandwidth of 20 kHz, namely the EVS codec (initials of the English words "Enhanced Voice Services" meaning "Enhanced Voice Services"). To allow voice calls in "HD Voice" or "Full-HD" quality, the calling and called party's telephones must contain compatible codecs compliant with these standards; the mobile networks where the two mobile phones are located must both be compatible with VoLTE technology and be interconnected by IP gateways that do not transcode the voice and that preserve the integrity of the IP streams containing the VoLTE frames. or ViLTE.

Compared with the High Definition Voice offered on some UMTS mobile networks, and by some proprietary VoIP technologies (such as "Skype") that are already compatible with "HD Voice" codecs, the VoLTE / ViLTE technique brings in addition 5 the low latency, limited jitter and debit guarantees offered by the LTE and its core network IMS, which contributes to improving the perceived quality and stability of telephone or videophone communications. In order to be able to specifically evaluate the speech quality of a connection, the invention proposes to isolate the positive / negative acknowledgments relating to the packets containing the speech frames, by not taking into account, in the case of an LTE network, positive and negative acknowledgments relating to the data packets conveyed in the "QCI 1" bearer.

In this respect, it is recalled that a virtual channel is called a bearer for the communication data associated with a certain service or a certain quality of service within a connection. As explained by A. Basir on the website http://4g-Ite-world.blogspot.fr, a bearer is a set of network parameters that define how this data is processed when it is conveyed over the network. The network can indeed process certain data in a particular way, and process the other data normally. For example, some data streams could benefit from a Guaranteed Bit Rate (GBR), while others could have a relatively low bit rate. When a terminal attaches to a network, it is assigned a bearer called "default", which is maintained as the terminal remains attached to the network. A default bearer has a non-guaranteed QoS ("best effort" in English). Each default bearer has an IP address.

3037461 12 The terminal can be assigned additional default bearers. Each default bearer has an IP address of its own. On the other hand, so-called "dedicated" bearers provide tunnels for one or more particular traffics (such as VolP or video). A dedicated 5 bearer acts as an additional bearer associated with a default bearer previously established. It does not require a clean IP address because only the default bearers have an IP address. LTE networks usually implement two default bearers and a dedicated bearer (according to the LTE standard, one can have up to 10 eight bearers). As shown in Figure 2, the default bearer "QCI5" is used for SIP signaling in the IMS network; it is associated with a "QCI1" bearer dedicated to VoLTE and ViLTE voice frames (the ViLTE picture frames are usually conveyed in the "QCI2" bearer). Another default bearer is used for the other 15 types of "data" traffic of the mobile terminal (for example, video, chat, email, or browser). An embodiment of the invention will now be described with reference to FIG. In this embodiment, the number of packets relating to the bearer QCI1 to be transmitted is compared with the number of these packets actually received. For a given direction of transmission (amount or downlink), two counters are defined, namely "Total_to_Transmit_x" and Total_Data_x ", where" x "takes a different value depending on whether the transmission link is upstream or downstream (for example x = UL for the up link and x = DL for the downlink). These counters are initialized to O. In decreasing order of preference, the non-reception rate will be calculated: 30 - in both directions of transmission, or 3037461 13 - only in the uplink, or - only in the downlink. With regard to the up link, for a number of successive packets transmitted by said mobile terminal, following each reception of an acknowledgment / non-reception of packet (ACK / NACK) sent to the mobile terminal by the eNode B ( LTE base station) to which the mobile terminal is attached: - if the acknowledgment is the last that can be issued for a certain packet to be transmitted, - if the acknowledgment is positive (ACK), the mobile terminal increments Total_to_Transmit_UL and Total_ULK_data, and - if the acknowledgment is negative (NACK), the mobile terminal increments only Total_to_transmit_UL, and - if the acknowledgment is not the last one to be issued for a certain packet to be transmitted, - if the Acknowledgment is positive (ACK), the mobile terminal increments Total_to_Transmit_UL and Total_Data_UL, and - if the acknowledgment is negative (NACK), nothing is done (the symbol "Nop" in Figures 3 and 4 means "we do nothing" ).

Indeed, in the latter event, and in accordance with the conventional mechanism of acknowledgments of the data link layer, it is known that the packet to be transmitted, which has not been received, will be re-transmitted at least once more, which subsequently generate at least one positive or negative acknowledgment for the transmitted packet.

Similarly, with regard to the downlink, for a number of successive packets received by said mobile terminal, following each transmission of an acknowledgment / non-reception of packet (ACK / NACK) sent to the eNode B by the mobile terminal: - if the acknowledgment is the last one that can be issued for a certain packet to be transmitted, - if the acknowledgment is positive (ACK), the mobile terminal increments Total_to_Transmit_DL and Total_Data_DL, and - if acknowledgment is negative (NACK), the mobile terminal increments only Total_to_Transmit_DL, and 5 - if the acknowledgment is not the last that can be sent for a certain packet to be transmitted, - if the acknowledgment is positive (ACK) , the mobile terminal increments Total_to_Transmit_DL and Total_Data_DL, and - if the acknowledgment is negative (NACK), nothing is done.

10 For a given link (amount or descendant), the non-reception rate (TNR) is deduced: TNR = 1- (Total_Data_x / Total_to_transm and_x). The value of TNR thus obtained, considered as an estimate of BLER, implies for a given type of codec (as explained above with reference to FIG. 1) a very precise value of the MOS, that is to say a well defined vocal quality of communication. It will now be explained with reference to FIG. 4 how the invention can be used to assist a multimode mobile terminal in deciding whether it is expedient for it to switch a telephone or videophone communication from a first access network. (eg LTE type) to a second access network (for example, WiFi type). To avoid untimely decisions, this decision support method will preferably take into account a predetermined, statistically significant number of transmitted packets (eg, 50 packets). Alternatively, the method can take into account a predetermined duration of communication (for example, 1 second), but in this case it is not known exactly how many packets will be taken into account because it will depend on the duration of the silence in the conversation! In FIG. 4, by way of example, the embodiment of the invention described above with reference to FIG. 3 is applied for 50 successive packets in the uplink (x = UL) or for 50 successive packets in the downlink (x = DL). If the TNR is greater than a certain threshold associated with the codec implemented by the mobile terminal, the latter switches from the LTE network to the WiFi network (if it is the only failover criterion configured in the mobile terminal). Indeed, when the TNR is above this threshold, the MOS is below a desired threshold. For example, it has been considered with reference to Figure 1 that the voice quality is insufficient when the MOS is less than 2.50.

The invention can be implemented within mobile terminals by means of software and / or hardware components. The software components can be integrated into a typical network node management computer program. Therefore, as indicated above, the present invention also relates to a computer system. This computer system conventionally comprises a central processing unit controlling signals by a memory, as well as an input unit and an output unit. In addition, this computer system can be used to execute a computer program having instructions for implementing any of the methods of assessing a communication quality or decision support according to the present invention. 'invention. Indeed, the invention also relates to a downloadable computer program from a communication network comprising instructions for carrying out the steps of a method of evaluating a communication quality or a method of assisting to the decision according to the invention, when executed on a computer. This computer program may be stored on a computer readable medium and may be executable by a microprocessor. This program can use any programming language, and be in the form of source code, object code, or intermediate code between source code and object code, such as in a partially compiled form, or in n any other desirable form. The invention also relates to an information carrier, irremovable, or partially or completely removable, readable by a computer, and including instructions of a computer program as mentioned above. The information carrier may be any entity or device capable of storing the program. For example, the medium may comprise storage means, such as a ROM, for example a CD ROM or a microelectronic circuit ROM, or a magnetic recording means, such as a hard disk, or a key USB ("USB flash drive"). On the other hand, the information medium may be a transmissible medium such as an electrical or optical signal, which may be conveyed via an electrical or optical cable, by radio or by other means. The computer program according to the invention can in particular be downloaded to an Internet type network. Alternatively, the information carrier may be an integrated circuit 20 in which the program is embedded, the circuit being adapted to execute or to be used in the performance of a method of evaluating a quality of communication or communication. a decision aid method according to the invention.

Claims (10)

REVENDICATIONS1. A method for evaluating the quality of a Voice over IP communication in an access network, characterized in that it comprises the following steps implemented by a mobile terminal: - taking into account positive and negative acknowledgments relating to to packets transmitted or received by the mobile terminal within a virtual channel dedicated to the voice, and - determining a rate of non-reception of said packets. 10 2. Evaluation method according to claim 1, characterized in that it further comprises a step of determining, on the basis of said non-reception rate and the type of codec used by said mobile terminal, a level of voice quality of communication. 15 3. evaluation method according to claim 1 or claim 2, characterized in that said access network is a LTE type network (Long Term Evolution). 20 4. Method for assisting the decision to switch between access networks, characterized in that it comprises the following steps implemented by a mobile terminal: - determining, for a telephone or videophone communication on an access network , a packet non-reception rate according to claim 1 or, respectively, a voice quality level according to claim 2, and - comparing said packet non-reception rate with a packet non-reception threshold. depending on the type of codec used by the mobile terminal, or, respectively, said voice quality level with a voice quality threshold. 3037461 18 5. Mobile terminal, characterized in that it comprises means for: - taking into account, when said mobile terminal is engaged in a Voice over IP communication on an access network, positive and negative acknowledgments relating to packets transmitted or received by the mobile terminal within a virtual channel dedicated to the voice, and - determining a rate of non-reception of said packets. Mobile terminal according to claim 5, characterized in that it further comprises means for determining, on the basis of said non-reception rate and the type of codec used by said mobile terminal, a voice quality level of Communication. A mobile terminal, comprising means for determining a packet non-reception rate according to claim 5 or, respectively, means for determining a voice quality level according to claim 6, when said mobile terminal is engaged on a communication. telephone or videophone on an access network, characterized in that it furthermore has means for comparing said packet non-reception rate with a packet non-reception threshold depending on the type of codec used by the terminal mobile, or, respectively, said voice quality level with a voice quality threshold. 25 8. evaluation method according to any one of claims 5 to 7, characterized in that said access network is a network type LTE (Long Term Evolution). 9. Non-removable or partially removable or removable data storage means having computer program code instructions for performing the steps of an evaluation method according to any one of claims 1 to 3, or a decision support method according to claim 4. Computer program downloadable from a communication network and / or stored on a computer readable medium and / or executable by a microprocessor, characterized in that it comprises instructions for carrying out the steps of a method evaluation system according to any one of claims 1 to 3, or a decision support method according to claim 4 when executed on a computer.