WO2009153528A1

WO2009153528A1 - Method for managing the quality of service in a mobile communication system

Info

Publication number: WO2009153528A1
Application number: PCT/FR2009/051169
Authority: WO
Inventors: Nasr Benali
Original assignee: France Telecom
Priority date: 2008-06-19
Filing date: 2009-06-19
Publication date: 2009-12-23

Abstract

The invention relates to a method for managing quality of service. The method includes, as novel features: a step for determining a representative parameter for a quality of service in the stationary portion of a first communication network; a step for verifying that at least one first predefined threshold has been reached by said parameter; and a step for changing a number of frames conveyed in a packet on said first network. A decrease in the throughput required of packets and a decrease in the header costs are thereby provided.

Description

METHOD FOR MANAGING QUALITY OF SERVICE IN A MOBILE COMMUNICATION SYSTEM

The invention relates to a quality of service management method. The invention also relates to a device that can be used in a communication network.

Quality of service management mechanisms are known from US 2005/0185651. This document addresses the problem of ensuring (or guaranteeing) the quality of service at the radio interface level (also called air interface) between a mobile terminal and a node-B, the base station in a "UTRAN" system (acronym "English" UMTS "(English acronym for" Universal Mobile Telecommunications System ") and" Terrestrial Radio Access Network "). The document describes a device called "Packet Scheduler" which decides which mobile terminal can exchange data with the node-B. In the case of a deterioration of the quality of service, the Packet Scheduler can simply decide to no longer ensure the traffic between a given mobile terminal and the corresponding node-B in order to increase the quality of service for the other terminals related to this node-B. Document US 2005/0185651 thus relates to the radio part of a mobile communication network. The radio part of a mobile communication network differs from the fixed part in terms of quality of service assurance, so that the QoS management mechanism (s) can be used on the radio part are not on the fixed part, which has a disadvantage. The document US 2005/0185651 therefore addresses the problems related to the quality of service on the radio interface, in contrast to the present invention. The invention proposes a solution that does not have the aforementioned drawbacks. The method according to the invention is thus characterized in that it comprises:

a step of determining a representative parameter for a quality of service in the fixed part of a first communication network,

a step of verifying that at least a first predetermined threshold has been reached by said parameter, and a step of modifying a number of frames transported in a packet on said first network. This advantageously makes it possible to reduce the required packet rate and reduce header costs. Example: to transmit the same data length (say 1000 ms of AMR (Adaptive Multi-Rate) is needed 50 packets if each packet contains 20 ms of voice (50 * 20 = 1000 ) but if we increase the size up to 40 ms, that is to say if we increase the number of frames, we will need only 25 packets (25 * 40 = 1000). In this case, the modification is therefore an increase in the payload of the transport packet, but the modification can also be a reduction in the payload of the transport packet, for example after an increase in the payload of the transport packet. In one embodiment, the method is characterized in that it comprises:

a step of detecting that at least a second previously defined threshold has been reached by said parameter,

a step of transferring at least one communication session from said first communication network to a second communication network. Advantageously, the session transferred to the second network will no longer be affected by the degradation of the transport medium of the first network and the session transferred to the second network releases bandwidth on the first network. The invention also relates to a device that can be used in a communication network, characterized in that it comprises: means for determining a representative parameter for a quality of service in the fixed part of a first network Communication,

means for verifying that at least a first predetermined threshold has been reached by said parameter, and

means for modifying a number of frames carried in a packet on said first network.

In one embodiment, the device is characterized in that it comprises:

detection means that at least a second predetermined threshold has been reached by said parameter,

means for transferring at least one communication session from said first communication network to a second communication network.

These devices have at least one of the advantages mentioned above. In one embodiment, the method is characterized in that said parameter belongs to the group comprising:

- a rate of packet loss,

- a routing period and - a variation of the said period.

In one embodiment, the device is characterized in that said device belongs to the group comprising:

- a so-called residential terminal and

a node called mobile concentration. The invention also relates to a computer program comprising instructions for implementing the method according to any one of the methods described above, when this program is executed by a processor. The invention also relates to a computer program product, comprising program code means, stored on a computer-readable medium, for implementing the method according to any one of the methods described above, when said program is running on a computer.

The invention will be better understood with the aid of the following description, given by way of nonlimiting example and with reference to the appended drawings, in which: FIG. 1 is a partial block diagram describing a communication system implementing the invention,

FIG. 2 illustrates a processing of the payload of an RTP packet (for the English expression: "Real time Transport Protocol") according to the invention, and

- Figure 3 is a timing diagram illustrating the method according to the invention. In the following, the invention will be explained on the basis of the universal mobile telecommunication system (in English: UMTS, acronym for "Universal Mobile Telecommunications System"). However, the invention can also be used in a different system, for example called "second generation" or "2G", as GSM (acronym for "global System for mobile communications", formerly "mobile special group") . FIG. 1 schematically represents a so-called "fixed / mobile convergence" telecommunication system 121. Such a telecommunication system 121 allows convergence between a mobile telecommunication network and a fixed telecommunication network. In the UMTS case, such a system 121 comprises a residential terminal UMTS 16 with a UMTS antenna 13 and a processing unit 15, an ADSL modem 17 (abbreviation of the English expression: "asymmetric digital subscriber line": in French "ligne Asymmetric Digital Subscriber Subscriber "), a DSLAM 19 (acronym for" Digital Subscriber Line Access Multiplexer "), a BRAS 111 (acronym for" Broadband Remote Access Server "), a node mobile concentrator 113 (in English: mobile node concentrator "), a mobile switching node 115 (in English: mobile switching center (acronym: MSC)), an RNC 117 (acronym for" Radio Network Controller "). ) and a base station 119 called node-B. In such a system 121, some elements may have more than one copy (for example the base station 119), but for the sake of simplification each element is represented only once in Figure 1. In addition, although in the fig Since each element of the system 121 is individually represented, it is possible to group the functionality of two (or more) elements into a single device. Indeed, each element shown in Figure 1 logically plays a specific function, and it is possible to physically integrate in a single device two logical elements.

For example, it is possible to integrate ADSL modem functions 17 with the UMTS 16 residential terminal. In this case, the UMTS 16 residential terminal will connect to the DSLAM 19 directly. The UMTS 16 residential terminal (integrating the UMTS antenna 13) is connected to the ADSL modem 17 via an Ethernet link.

The ADSL modem 17 is linked to the DSLAM 19 via an ADSL link. The ADSL modem 17 transforms the digital signal received from the UMTS 16 residential terminal into a transportable signal on this ADSL link, and also carries out the inverse transformation. A PPP link exists between the ADSL modem 17 and the ARM 111. The ARM 111 is then interconnected via a link. IP connection (for: "Internet Protocol"), also known as "IP backbone", to the mobile concentration node 113.

The mobile concentration node 113 is a platform, that is to say one or more equipment (s) co-located (s) for performing additional functions (security, provisioning, management of the equipment, RAN functions (for, in English: "Radio Access Network"). It makes it possible to aggregate the IP connections of the different UMTS residential terminals that it manages in its location; allows the connections to the mobile core network through the standard interfaces called Iu-CS and Iu-PS and D '/ Gr' (known to those skilled in the art), and performs functions of RAN type such as the triggering of notices of paging, or mobility procedures between the cell constituted by the UMTS 16 residential terminal and the cells associated with the RNC equipment constituting the UMTS network (macro, see below) and superimposed on the residential cell. The other links between the various elements of the system 121 are well known to those skilled in the art and we will not explain them in more detail.

The mobile switching node 115 is in charge of managing the circuit switched services (e.g., voice communications) of the mobile terminal 11 roaming within its coverage area. In summary, the UMTS 16 residential terminal establishes a secure IP connection via the aforementioned elements with the mobile concentration node 113. This connection will convey the data of the mobile subscriber (and the signaling) of the residence of the mobile subscriber at the mobile concentration node 113.

A radio network is associated with the UMTS 16 residential terminal. The mobile terminal 11 can select this radio network and register there. This network will be called in the following "residential" radio network. This is to distinguish with the network associated with conventional B-nodes which, too, constitute a radio network. This radio network is also called "public" UMTS network or "macro" UMTS network. The mobile terminal 11 can also select and register to the latter radio network. The two networks - residential and public UMTS network - are two parallel radio networks. The mobile telephone 11 will generally be connected to the public UMTS network but may select the "residential" radio network as soon as it is under the radio coverage of the terminal UMTS 16. In fact, when the user of the mobile phone 11 goes home, settings of the radio network can force the mobile phone to cling to the residential terminal UMTS 16 even if the radio conditions of the service cell - public cell or macro - are better. Unlike the public UMTS network, the residential network allows the user of the mobile phone 11 to benefit from a wider range of services for a lower price, because when the mobile phone 11 has registered at the terminal UMTS 16, it uses the ADSL link as well as the Internet protocol-based infrastructure instead of the private infrastructure of the mobile operator (notably the B-nodes 119 and the RNCs 117, a set known as English name of "UTRAN") for the transport of the user's traffic (eg voice frames, data packets, signaling).

The word "convergence", see above, is used in the sense that the IP network infrastructure, for example through a conventional ADSL connection, is used in place of a private infrastructure of a UMTS mobile operator ( which links the base station (s) 119 called "Node (s) B" to the RNCs 117) when a UMTS mobile subscriber enters his residence and his mobile telephone 11 hooks up to the UMTS residential terminal 16 The data (for example, the voice) of the mobile service is conveyed on the public IP infrastructure when the user enters his home and connects to the UMTS 16 residential terminal (itself connected to a conventional ADSL link for example) .

However, the ADSL link usually belongs to an ADSL access provider. Such a provider may very well be a company independent of the mobile operator. However, the ADSL access provider does not necessarily implement a quality of service management mechanism in its ADSL / IP network and therefore the delay-sensitive traffic and the loss of data such that the voice may be altered. for example due to congestion in the IP transport media. To ensure a certain level of quality for UMTS mobile services, the mobile operator will therefore have to set up quality of service management mechanisms independently of the possible quality of service processing mechanisms deployed in the ADSUJP network. Such a mechanism is all the more necessary that the UMTS service is not necessarily the only service that passes through the ADSL modem 17. In fact, it is possible that the UMTS service coexists with other services (called "native Internet"). "), for example the consultation of web pages, the remote downloading of a file from a server via a personal computer 123 or the broadcasting of the television on the Internet to a television 125. All these services share the same ADSL link which however has a limited bandwidth. Each service sends data packets across the ADSL link. The ADSL bandwidth can therefore be reached quickly, for example when using several services simultaneously, such as downloading files (particularly a large file), a television transmission through ADSL and several UMTS voice communications. . In such a situation, the possibility of congestion becomes very likely both at the level of the ADSL modem 17 and further in the network at the level of the DSLAM 19 for example. Congestion can introduce a significant delay in the routing of packets and / or can lead to loss of packets. This produces a deterioration of the quality of service (in English: QoS for "Quality of Service"). In telecommunication system 121, there are two potential congestion points (among others). A first potential point of congestion is the ADSL modem 17 in the upstream direction of the traffic. Another potential congestion point is ARM 111 in the downstream direction of traffic.

In the case where no separation mechanism (and processing) of flows or QoS management at the ADSL / EP level is implemented, UMTS service packets may be altered in the same way as another service, eg Internet service. Therefore, in the event that congestion occurs at the ADSL / IP network, a user of the UMTS service who has for example a voice communication in progress, will see his communication deteriorate. It should be noted that voice communication is particularly sensitive to delays and packet loss. The voice frames are sent encapsulated in so-called RTP (Real-time Transport Protocol) packets between the UMTS residential terminal 16 and the mobile concentration node 113 of the telecommunication system 121. The acceptance of a voice communication in the UMTS residential kiosk system results in the establishment of a RTP session between the UMTS 16 residential kiosk and the MTS 113. If multiple UMTS voice calls are requested simultaneously (eg 4 This means that there are several RTP sessions in progress (4 RTP sessions).

As shown in FIG. 2, such an RTP packet 211 generally comprises a header 21 and a payload. 22. As will be known to those skilled in the art, RTCP reports (for the expression "Real-time Transport Control Protocol") make it possible to check the state of an RTP session such as the RTP session between the residential terminal 16 and the mobile concentrator node 113, particularly the loss of RTP packets belonging to this session and the delays of their arrival (from the residential terminal UMTS 16) to the mobile concentration node 113 (and vice versa). These reports can be inspected by the UMTS 16 residential terminal and the mobile concentration node 113. Thus, it is possible, by RTP session, to detect the RTP packet losses and the variations in the routing delays of these packets from the UMTS 16 residential terminal to the mobile concentration node 113 (and vice versa). In addition, it is also possible to check the status of the queues (in English "buffer") of the variation of the delay (in English "jitter") of the packets of the same session RTP (at the level of the node of concentration mobile 113) and thus to determine that said packets have undergone a serious variation of delay which could alter the voice communication in progress. If the reports indicate a high percentage of RTP packet loss, or excessive routing delays or with large variations, a treatment according to the invention will be applied. Before the treatment can be performed, the definition of treatment trigger points is necessary. By way of example, the following two thresholds can be considered: the RTP packet loss threshold (of value of 1%, for example) and the maximum delay threshold (of value of 170 ms, for example). If the percentage of packet loss exceeds 1% or the routing delay exceeds 170 ms, the processing according to the invention will be carried out. This processing, returning to FIG. 2, comprises an increase in the payload 22 transmitted in the RTP packet 211. In the solutions currently proposed by the constructors of the UMTS residential terminals, there is found as payload 22 a frame 23 of 20 ms of voice ( or video) in an RTP packet. The processing according to the invention consists in including a second frame 25, and, if necessary, a third frame 27 and a fourth frame 29 of 20 ms each, in the same packet RTP 211. In the case where the payload includes two frames of 20 ms, 40 ms of voice are sent instead of two RTP packets including 20 ms of voice. This will halve the bit rate required for one or more RTP sessions, depending on the number of frames multiplexed in the same RTP packet (for example: at 20 ms frame, 50 RTP packets must be transmitted against 25 RTP packets if 40 ms of frames are multiplexed in the same RTP packet). Halving the rate of multiple RTP sessions on the same ADSL access would reduce the overall loss rate and therefore could resolve congestion and restore quality of service. However, this multiplexing of two, three, or four voice frames in the same RTP payload does not always mean restoration of the quality of service. Indeed, in cases of severe congestion on the IP network, the packet losses (or the delay of their routing) can be large enough to impact the quality of the session (sessions) voice despite the multiplexing processing described herein. -above. In such a situation, the transfer of the voice session (for example the voice session which is the most impacted) must be made to the UMTS macro network if the terminal 11 is under the coverage of the UMTS macro network in order to maintain the quality of the voice. the communication in progress. In this case, the QoS will be restored (no more frame loss or delay) for the forwarded voice session, and the bandwidth of the ADSL link will be freed from the required value for a voice session (approximately 80 kbps when the IP link is secured by IPSEC technology (the acronym for "Internet Protocol SECurity"). This would have a positive effect on the current RTP sessions on the same ADSL access. Specifically, the remaining RTP sessions could regain a normal quality of service by releasing the bandwidth (because there will be a decrease in the overall loss rate) after the transfer of the voice session to the macro network. In summary, according to the invention, the size of the payload transmitted in an RTP packet is increased in steps of 20 ms. The limit is to transmit up to 80 ms of voice in the same payload of the RTP packet (that is 4 frames of 20 ms) and to make a control on the RTCP reports of this session to each increase to observe improvement or degradation of the quality of service. If the quality remains bad even when including 80 ms of voice in the same RTP packet, a transfer of the session to the mobile macro network is performed.

The method according to the invention will be explained in more detail with reference to the timing diagram of FIG. 3. The RTP packet loss rate will be used as a triggering parameter. However, the method may also use as triggering parameter the routing delay of the RTP packets or the delay variation. In step 30 of definition, there is the definition of the following thresholds:

RTP_Loss_sampie_size_Modify: This threshold triggers the increase of the size of the payload transmitted in a RTP packet (for example an increase of 20 ms of additional voice in the same RTP packet) on detection of a consequent increase in the rate of loss of packets on a given session.

RTP_Loss_Sampie_size_Modify_Desactivate: this threshold allows the triggering of the reduction of the size of the payload transmitted in an RTP packet (for example a reduction of 20 ms of voice in a RTP packet) on detection of a consequent decrease of the rate of packet loss on a given session.

RTP_Loss_Redirection_Macro: this threshold makes it possible to trigger the transfer of the session on the UMTS macro network on detection of a consequent increase in the rate of loss of packets on a given session.

In addition, in step 30, there is a step of identifying a target macro UMTS cell to which all voice sessions experiencing significant degradation on the network of UMTS residential kiosks will be transferred. In order to identify this cell, when the mobile subscriber branches his UMTS 16 residential terminal on his ADSL modem 17 and on again, said terminal contains devices UMTS radio transmitter / receiver for scanning the local radio environment and detect the macro cells that cover the residence of the subscriber. The cell that will be registered in the UMTS residential terminal is the UMTS cell having the highest radio field level among the detected cells. It will then be used for voice session transfers from the UMTS residential kiosk network to the macro network. This transfer is conventionally known as "Blind Handover". During a determination step 31, the method determines the rate of RTP packet loss through the RTCP reports. This step 31 will take place in the UMTS 16 residential terminal and / or in the mobile concentration node 113.

During a first verification step 33, the method checks whether a threshold for the RTP packet loss rate has been reached. If a threshold is not reached, the process returns to step 31 of determination. Otherwise, during a second verification step, the method verifies whether it is a threshold overflow in the upstream direction or the opposite in the downward direction.

If it is an ascending overshoot during a detection step 371, the method checks whether the RTP_Loss_Red_Macro threshold has been crossed, that is to say if the loss rate has increased in a sufficiently high manner. to exceed the trigger threshold RTP_Loss_Loss_Redirection_Macro.

If this is the case, the method continues to step 38 of checking macro network coverage.

If this is not the case, the method continues to a step 37 of verifying the maximum payload. During step 37, the method checks whether the maximum payload for the packets

RTP has been reached. For example, in the case described above, the method therefore checks whether the RTP packets already contain 80 ms of voice in their payload.

If this is the case, the process continues to step 31. If it is not, the process continues to step 39 of increase. In step 39 of increase, the size of the payload is increased (an additional frame is included) for example of 20 ms. After step 39, the process continues to step 31.

If, in step 35, it was not an ascending overflow but rather a decrease in the loss rate, in the reduction step 311, the size of the payload is reduced for example by 20 ms.

After step 311, the process continues to step 31.

In step 38 of checking macro network coverage, the method verifies whether the terminal 11 is under the coverage of the macro network (in other words if there is an identification of the target public cell defined in FIG. step 30. If this is not the case, the process continues to step 31.

If, on the other hand, the terminal 11 is under the coverage of the UMTS macro network, the method, at the transfer step 313, transfers the voice session of the network of the UMTS residential terminals to the macro network. It should be noted that FIG. 3 represents the processing of a voice session. After transferring this session, the process stops for this session. But it continues for the remaining sessions, if there are any.

In the foregoing, the idea according to the invention has been explained from UMTS. However, the invention is reusable on any system having an IP link (or, in English "IP bearer") between the residential terminals and the mobile concentration node 113.

In, for example, the GSM case, the UMTS 16 residential terminal will be a "residential" BTS terminal and the mobile concentration node 113 a specific BSC controller that manages it. During step 30, there will be an identification of a target macro GSM cell. It is also conceivable to use a GSM transmitter / receiver device in the UMTS residential terminal. In this case during step 30, two cells will be recorded in the residential terminal: a GSM cell and a UMTS cell. The mobile terminal can thus make the transition from the UMTS residential cell to a UMTS cell or a GSM cell.

Claims

1) quality of service management method characterized in that it comprises:

a step (31) of determining a representative parameter for a quality of service in the fixed part of a first communication network,

a step (33) of verifying that at least a first previously defined threshold has been reached by said parameter, and

a step (39, 311) of modifying a number of frames carried in a packet on said first network.

2) Method according to claim 1 characterized in that it comprises:

a step (371) of detecting that at least a second predetermined threshold has been reached by said parameter,

a step (313) of transferring at least one communication session from said first communication network to a second communication network.

3) Device adapted to be used in a communication network characterized in that it comprises:

means for determining a representative parameter for a quality of service in the fixed part of a first communication network,

4) Device according to claim 4 characterized in that it comprises:

means for transferring at least one communication session from said first communication network to a second communication network. 5) Method according to any one of claims 1 to 2 characterized in that said parameter belongs to the group comprising:

- a rate of packet loss,

- a routing period and - a variation of the said period.

6) Device according to any one of claims 3 to 4 characterized in that said device belongs to the group comprising:

a so-called residential terminal (16) and a so-called mobile concentration node (113).

7) computer program comprising instructions for implementing the method according to any one of claims 1 to 2, when the program is executed by a processor.

8) A computer program product, comprising program code means, stored on a computer-readable medium, for implementing the method of any one of claims 1 to 2, when said program is operating on a computer program medium. computer.