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WO2001078430A1 - Application de rtp et de rtcp dans le transport amr sur reseaux voix sur ip - Google Patents

Application de rtp et de rtcp dans le transport amr sur reseaux voix sur ip Download PDF

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Publication number
WO2001078430A1
WO2001078430A1 PCT/EP2000/003233 EP0003233W WO0178430A1 WO 2001078430 A1 WO2001078430 A1 WO 2001078430A1 EP 0003233 W EP0003233 W EP 0003233W WO 0178430 A1 WO0178430 A1 WO 0178430A1
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WO
WIPO (PCT)
Prior art keywords
codec
amr
network element
stream
network
Prior art date
Application number
PCT/EP2000/003233
Other languages
English (en)
Inventor
Sami Kekki
Original Assignee
Nokia Corporation
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Nokia Corporation filed Critical Nokia Corporation
Priority to AU2000244000A priority Critical patent/AU2000244000A1/en
Priority to PCT/EP2000/003233 priority patent/WO2001078430A1/fr
Priority to EP00925202A priority patent/EP1275261A1/fr
Publication of WO2001078430A1 publication Critical patent/WO2001078430A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W88/00—Devices specially adapted for wireless communication networks, e.g. terminals, base stations or access point devices
    • H04W88/18—Service support devices; Network management devices
    • H04W88/181—Transcoding devices; Rate adaptation devices
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer

Definitions

  • the present invention relates to a method and a network system for conveying an Adaptive Multi-Rate (AMR) codec stream over multimedia IP networks.
  • AMR Adaptive Multi-Rate
  • the invention relates to transmitting an Adaptive Multi- Rate (AMR) codec stream over networks.
  • a network can be a multimedia IP network, a third generation (3G) mobile network or a VoIP (Voice over IP) network in general .
  • 3G third generation
  • VoIP Voice over IP
  • an AMR speech coding scheme has been selected. It is also to be used by the GSM (GPRS/EDGE GSM) .
  • GSM GPRS/EDGE GSM
  • the UMTS Release 2000 is targeted to provide IP service support over the UMTS air interface (Uu) .
  • a transmission channel between two codecs (coder-decoder) in a network has a fixed data rate.
  • codecs coder-decoder
  • two mobile stations are connected with each other via a mobile network.
  • the first mobile station contains a first codec
  • the second mobile stations contains a second codec.
  • the codecs perform the necessary encoding/decoding for converting a voice signal into a digital signal which is transmitted via the network and vice versa.
  • the codecs serve to provide a certain data rate, for example, 22.8 kbit/s on a so-called “full rate channel”. In case of AMR codecs, it is possible to switch this data rate to another data rate, for example, to 11.4 kbit/s on a so- called “half rate channel”. This switching between the different data rates has to be performed simultaneously by both of the codecs involved.
  • control of the AMR codecs involved is important.
  • the control of the AMR codec is UMTS specific (ref. e.g., to 3GPP TS25.415) .
  • the object underlying the present invention resides in providing a method by which an improved control of AMR codecs is possible.
  • This object is solved by a method for conveying an Adaptive Multi-Rate (AMR) codec stream between a first codec and a second codec over a network, comprising the steps of transmitting AMR frames via a Real-Time Transport Protocol (RTP) stream, and transmitting an AMR control signal via a Real-Time Transport Control Protocol (RTCP) stream.
  • AMR Adaptive Multi-Rate
  • a control channel is provided between the two peer AMR codecs in order to benefit from the capabilities of the AMR concept.
  • the codecs have a mechanism to signal to its peer the requests to change the codec mode.
  • the macrodiversity needs to be taken into account in the AMR control, as all branches of the call need to have the same coding mode. Since according to the invention the coding mode, i.e., the AMR frames, are the same during the transmission even in different networks, this condition can be fulfilled by the invention.
  • WCDMA Wideband CDMA, Wideband Code Division Multiple Access
  • the interfaces are re-usable between different application areas.
  • the number of different interfaces (protocol stack, procedures) in UMTS or in a Voice over IP environment (wireless and/or fixed) can be minimized in order to avoid interworking problems and unnecessary implementation effort.
  • a first network element in which the first codec may be arranged, can be a mobile station, the method further comprising the step of performing a handover from a first base station to a second base station, wherein each base station is adapted to determine a need for a codec mode change and to suggest a codec mode independently and the mobile station is adapted to decide of the codec mode change on the basis of a predetermined criterion.
  • the predetermined criterion can be using the lower quality mode of the codec modes suggested by the base stations.
  • the predetermined criterion can be using the higher quality mode of the codec modes suggested by the base stations. By this measure, a high quality connection can be achieved.
  • the first network element can be a mobile station, the method further comprising the step of performing a handover from a first base station to a second base station, wherein a network control element is adapted to determine a codec mode change.
  • a central network element e.g., a Serving Radio Network Controller SRNC
  • SRNC Serving Radio Network Controller
  • a codec mode which has been determined by the network control element as described above can be sent to the first base station and/or the second base station via the Real-Time Transport Control Protocol stream.
  • the invention also proposes a network element comprising a frames extracting/inserting means which is adapted to insert and/or extract Adaptive Multi-Rate (AMR) frames from and/or into a Real-Time Transport Protocol (RTP) stream, and a control signal extracting/inserting means which is adapted to insert and/or extract an AMR control signal from and/or into a Real-Time Transport Control Protocol (RTP) stream.
  • the network element can be a mobile station, a gateway or the like.
  • the network element can comprise a mode deciding means adapted to determine a need for a codec mode change and to suggest a codec mode .
  • the network element can further comprise a mode deciding means adapted to decide, in case of a handover between a first base station and a second base station, a codec mode change on the basis of received suggested codec mode changes and of a predetermined criterion.
  • the predetermined criterion can be using the lower quality mode of the codec modes suggested by the base stations, or, alternatively, using the higher quality mode of the codec modes suggested by the base stations.
  • the mode deciding means can be arranged in a further network element (SRNC) .
  • SRNC network element
  • a network system comprising a first network element including a first codec for generating Adaptive Multi-Rate (AMR) codec frames, a second network element (BS) for receiving said AMR frames and including a transmitting means (CCU) for transmitting said AMR frames via a Real-Time Transport Protocol (RTP) stream and for transmitting an AMR control signal via a Real-Time Transport Control
  • AMR Adaptive Multi-Rate
  • BS second network element
  • CCU transmitting means
  • RTP Real-Time Transport Protocol
  • RTCP Transmission Control Protocol
  • GW third network element
  • AMR frames from said RTP stream
  • AMR control signal from said RTCP stream and for outputting said AMR frames and said AMR control signal
  • fourth network element for receiving and processing said AMR frames and said AMR control signal.
  • the first and the second network elements can be separately arranged network elements, or, alternatively arranged within a single network element.
  • the third and fourth network elements can be separately arranged network elements or, alternatively, be arranged in a single network element.
  • Fig. 1 shows a block diagram of a network configuration in which a method according to an embodiment is applied
  • Fig. 2 shows a protocol stack for the transport within the network shown in Fig. 1;
  • FIG. 3 shows a practical implementation of the embodiment shown in Fig. 1;
  • Fig. 4 shows the situation of a handover between two base stations according to a second embodiment
  • Fig. 5 shows the situation of a handover between two base stations according to a third embodiment
  • Fig. 6 shows an example for a gateway without a codec.
  • the method according to the invention uses the so-called Real-time Transport Protocol (RTP) to transmit AMR frames via a network, and the Real-time Transport Control Protocol (RTCP) is applied to transmit corresponding AMR control commands necessary for the codecs involved via the network .
  • RTP Real-time Transport Protocol
  • RTCP Real-time Transport Control Protocol
  • the Real-time Transport Protocol (RTP, RFC1889) is a widely-used transport protocol for any real time communications over IP. There are already many payload types defined for RTP in order to support specific real time applications (like H.261 video, G.721 audio, etc.).
  • RTCP Real-Time Control Protocol
  • UDP port number i.e., the User Datagram Protocol port number
  • UDP port number i.e., the User Datagram Protocol port number
  • the invention is to apply RTCP for end-to-end codec control (i.e., control between the peer codecs) in UMTS and in whatever environment the AMR is used. It is noted that a narrow-band or a wide-band AMR can be used. As a consequence, there is a need to specify an RTP payload type for AMR coded speech as well as the mode change support into RTCP.
  • a User Equipment On the left side, a User Equipment (UE) is shown which comprises an AMR codec (coder-decoder) for converting source signals like speech signals into data signals having a particular data rate.
  • the AMR codec is adapted to form the output data signals into AMR frames, i.e., to transmit these data signals in an AMR frame stream or AMR frame channel .
  • the necessary control signals for signalling changes of the codec mode and so on to a counterpart codec on the receiving side i.e., the AMR control signals are transmitted via an AMR control stream or AMR control channel.
  • the AMR control signals include AMR mode commands.
  • the counterpart codec resides in this case in a Media Gateway (MGW) . That is, the MGW may contain the codec function (physically the transcoder) . In general, the MGW performs functions like media conversions from one format to another (e.g., from an IP network to a Public Switched Telephone Network (PSTN)), payload processing (speech transcoding (i.e., the function of a codec) , echo cancelling etc.) .
  • PSTN Public Switched Telephone Network
  • the AMR frames and the AMR control signals are transmitted via the UMTS Uu interface to the Radio Access Network (RAN) .
  • the AMR frames and the AMR control signals are received by a Base Station (BS) .
  • BS Base Station
  • the AMR frames which have been transmitted from the UE to the BS are inserted into an RTP stream, and the AMR control signals which have been transmitted from the UE to the BS are inserted into an RTCP control stream.
  • the Iu interface can be the lu-PS interface between the RAN and the Packet Switched (PS) Core Network (CN) .
  • PS Packet Switched
  • CN Packet Switched Core Network
  • the invention is also applicable for the Iu-CS interface between a Circuit Switched (CS) Core Network (CN) .
  • the RTP stream and the RTCP stream are transmitted via an interface lu-PS (also defined in UMTS) to a Serving GPRS Support Node (SGSN) . Both streams are transmitted via an interface Gn (also defined in UMTS) to a Gateway GPRS Support Node (GGSN) . Finally, the RTP stream and the RTCP stream are transmitted via an interface Gi to a Gateway (MGW) which comprises a codec.
  • the gateway is adapted to extract the AMR frames from the RTP stream and to extract the AMR control signals from the RTCP control stream.
  • the codec of the gateway converts the received AMR frames into the data signals as needed for further transmitting or processing in the network to which the gateway is connected. Furthermore, the codec is adapted to change its mode in response to the AMR control signals received in the RTCP control stream.
  • Fig. 2 the corresponding protocol stack for the transport outlined above is shown. It is noted that this protocol stack is to be defined for the UMTS Release 2000 (R00) terrestrial transport, i.e., the transport between the Base Station and the GW node (i.e., the node where the codec resides in the network side) .
  • the user data part of the stack presented in Fig. 2 can be e.g., AMR speech frames conveyed in a header compressed RTP/UDP/IP payload.
  • the header compression/stripping has been standardized in UMTS to take place in the Mobile Station (MS, i.e., the UE) and in a Radio Network Controller (RNC, as a part of the RAN) .
  • MS Mobile Station
  • RNC Radio Network Controller
  • the UE as shown in Fig. 1 is a Mobile Station MS.
  • the mobile station MS comprises a codec including a Channel Coding Unit CCU to be described later.
  • The. MS is connected via a radio connection with a Base Station BS .
  • the base station BS also comprises a Channel Coding Unit CCU.
  • the CCU serves as a frames and control signal inserting/extracting means.
  • an AMR frame is inserted into the RTP frame payload as such.
  • the Channel Coding Unit CCU performs the radio channel coding (downlink) and decoding (uplink) . That is, in the case of an uplink, the CCU receives the AMR frame and the codec mode information (i.e., AMR control signal) for the uplink radio frame structure which has been transmitted from the UE to the BS . It then passes this information to a RTP protocol entity and to an associated RTCP protocol entity, respectively.
  • the base station BS is connected to a Radio Network Controller (RNC) .
  • RNC Radio Network Controller
  • the base station BS and the RNC are parts of the RAN (Radio Access Network) shown in Fig. 1.
  • the RNC is connected with a Gateway GW which comprises a further codec. It is assumed that the AMR frames transmitted between the mobile station MS and the gateway GW and vice versa are speech frames.
  • other kind of traffic can be transmitted, for example video data and other data which require a real-time treatment or have to be delivered quickly for other reasons .
  • the Channel Coding Units which are included in the MS and the BS, serve to determine the radio interface conditions, the resulting frame quality and the like, and serve to set different codec modes on the basis of these determination results.
  • the CCU performs, e.g., the convolutional coding/decoding or Turbo coding/decoding for the radio interface transmission. That is, the CCUs set the corresponding AMR codec mode commands into the AMR control signals.
  • the CCU inserts the commands, i.e., the AMR mode commands, into the RTCP stream.
  • the AMR codec change command for the downlink (DL) AMR stream (CHANGE DL MODE) is received over the radio interface, i.e., the interface between the MS and the BS .
  • a transmitting of a change command for the downlink AMR stream is indicated by the dotted arrow directed from the MS to the BS and further to the GW.
  • the receiving BS inserts the information into the RTCP message that it subsequently sends towards the network node where the codec resides (which is in Fig. 3 the GW) .
  • the codec mode is changed accordingly (the requested mode is included in the command) .
  • Each AMR frame carries the information of the used codec mode.
  • the change of codec mode is generally based on the radio interface conditions and the resulting frame quality that is determined by the Channel Coding Unit (CCU) located in the base station BS (UL) and in the mobile station MS (DL) .
  • the CCU of the base station BS performs such a determination
  • the mobile station MS performs the determination.
  • the change command only needs to be sent over the radio interface, provided that there is only one base station participating in the call.
  • the change command for the uplink AMR stream (CHANGE UL MODE) is indicated by a dotted arrow directed from the base station BS to the mobile station MS.
  • the mechanism for UL codec mode change may be different.
  • two alternative procedures are described as a second and a third embodiment, respectively.
  • Fig. 4 shows a situation, in which a (soft) handover between two base stations BS1 and BS2 according to the second embodiment is illustrated.
  • the mobile station MS comprises a codec and a CCU
  • the base stations BS1 and BS2 each comprise a
  • each base station involved in the given AMR call determines the need for UL codec mode change independently from each other.
  • the mobile station MS receives from the base station BS1 a command CHANGE UL MODE1 and from the base station a command CHANGE UL MODE2.
  • the MS is assigned the responsibility to make the decision of the mode change according to predefined criteria.
  • the CCU of the MS is adapted to decide which one of the received codec change command is to be applied by the codec.
  • the current codec mode is defined by the base station BS1.
  • the base station BS2 requests another codec mode by issuing the command CHANGE UL M0DE2 via the LI layer.
  • the mobile station MS has to decide which one of the modes should be applied, that is, whether the current mode should be changed to the new mode as requested by the base station BS2 , or whether the current codec mode should be maintained.
  • a criterion can be, e.g., to apply the current mode if it is better than the proposed new mode. That is, the CCU of the MS selects the codec mode which provides a better quality. For example, in case the current codec mode is full rate and the new mode requested by the base station BS2 is half rate, the CCU decides to maintain the full rate since the full rate provides a better quality.
  • An alternative criterion can be to apply the lower quality mode of the two. That is, the CCU of the MS selects the codec mode which has the lowest data rate. For example, in case the current codec mode is full rate and the new mode requested by the base station BS2 is half rate, the CCU decides to apply the half rate since the half rate has the lowest quality.
  • this criterion aims for keeping the branch active, at the cost of overall quality of the call.
  • a criterion can be to apply that codec mode which provides this certain data rate.
  • the UL codec mode change is determined by the Serving RNC (SRNC) of the calls by some defined algorithm.
  • SRNC Serving RNC
  • the CHANGE UL MODE command needs to be conveyed in the RTCP message (i.e., within the RTCP stream) to the SRNC from the base station that requests the codec mode change. Then the decision of the change is signaled via RTCP back to all involved base stations .
  • Fig. 5 This principle is illustrated in Fig. 5.
  • the situation is shown in which a handover from the base station BS1 to the base station BS2 takes place.
  • the mobile station MS is not involved in the decision on which codec mode is to be applied.
  • the CCU of the base station BS1 and the CCU of the base station BS2 transmit requests for a change of the UL mode to a SRNC (Serving RNC) .
  • the requests are in the form of AMR control signals which are transmitted via RTCP.
  • the AMR control signal contain also the necessary information (e.g., radio interface conditions and the like) .
  • the RTP streams are omitted and only the RTCP streams are shown.
  • the SRNC performs a mode decision, for example by use of a particular control unit on the basis of the received control signals. Thereafter, the SRNC inserts the decided mode, which is an AMR control signal (AMR mode command) into the RTCP stream by which the decided codec mode is transmitted to the base stations BS1 and BS2. In turn, the base stations BS1 and BS2 transmits the decided codec mode (i.e., the decided UL mode) to the mobile station MS.
  • AMR control signal AMR mode command
  • the reason for the codec mode change can be something else than the radio interface conditions. It can be used e.g., for congestion avoidance. In this case the same principles apply as have been defined in the above text. That is, the RTCP is used as the control channel for indicating the need for the codec mode change. It is noted that in the embodiments described above, the RTP and RTCP protocols do not have to be terminated in the network element where the codecs are located. They can be applied also only in some parts of the route between the first codec and the second codec. In Fig. 3, for example, the codecs are in the UE (i.e., the MS) and in the GW, whereas the RTP/TCP is used only between the BS and the GW.
  • the involved entity is the Channel Coding Unit (CCU) or an equivalent, that performs the radio channel decoding in the receiver.
  • CCU Channel Coding Unit
  • the RTP and RTCP are introduced.
  • the CCU After having decoded the speech frame and the change-DL-mode information from the cannel coded radio frame (the radio frame may have been convolutionally coded, for example) the CCU passes this information to RTP (speech frame) and to RTCP (change mode information) . After that, the RTP/RTCP convey this information.
  • the RTP becomes terminated (either in the second node, i.e., the destiny, or before that in some Gateway or equivalent) , the information needs to be taken out from RTP/RTCP and re-packed accordingly.
  • the RTP/RTCP stream would become terminated in the second BS and its CCU would take the change-DL-mode information and insert it into the radio frame along with the speech frame (from RTP payload) .
  • TrFO Transcoder Free Operation
  • Fig. 6 shows a further example in which the Gateway GW on the left side performs only the functionality of extracting and inserting of AMR frames into the RTP stream and the functionality of extracting and inserting AMR control signals in the RTCP stream. That is, this Gateway is an example for a network element in which no codec is included and which does not perform any decision regarding the codec mode or the like. This gateway merely extracts and/or inserts AMR frames into or from the RTP stream and extracts and/or inserts AMR control signals into or from the RTCP stream.
  • the RTP and RTCP streams are conveyed via an IP network (i.e., an "IP island") to a Media Gateway MGW which comprises a codec.
  • IP network i.e., an "IP island”
  • MGW Media Gateway MGW which comprises a codec.
  • the original signal e.g., a voice signal
  • a circuit switched network e.g., a circuit switched network to an opposite communication party.
  • the embodiments can be arbitrarily combined.
  • the second and the third embodiment can be combined. This can be effected with respect to the type of the mobile station concerned. That is, if the mobile station is capable of determining and suggesting an appropriate AMR codec mode, and if it also is capable to decide between different codec modes which are suggested by base stations involved in a handover, the method according to the second embodiment can be applied.
  • the method according to the third embodiment can be applied. That is, only in this case a central network element performs the decision.
  • a codec is arranged in a mobile station, and a further codec is arranged in a gateway.
  • this structure can be varied arbitrarily.
  • the gateway can be replaced by a second mobile station which is connected to the network via a second base station.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

L'invention concerne un procédé servant à transporter un flux de codec à débit multiple adaptatif (AMR) entre un premier codec et un deuxième codec sur un réseau. Le procédé comporte les étapes consistant à transmettre des trames AMR par l'intermédiaire d'un flux de protocole de transport en temps réel (RTP), et à transmettre un signal de commande AMR par l'intermédiaire d'un flux de protocole de commande de transport en temps réel (RTCP). Cette invention permet une régulation aisée du flux de codec AMR via des réseaux UMTS ou IP.
PCT/EP2000/003233 2000-04-11 2000-04-11 Application de rtp et de rtcp dans le transport amr sur reseaux voix sur ip WO2001078430A1 (fr)

Priority Applications (3)

Application Number Priority Date Filing Date Title
AU2000244000A AU2000244000A1 (en) 2000-04-11 2000-04-11 Application of rtp and rtcp in the amr transport in voice over ip networks
PCT/EP2000/003233 WO2001078430A1 (fr) 2000-04-11 2000-04-11 Application de rtp et de rtcp dans le transport amr sur reseaux voix sur ip
EP00925202A EP1275261A1 (fr) 2000-04-11 2000-04-11 Application de rtp et de rtcp dans le transport amr sur reseaux voix sur ip

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PCT/EP2000/003233 WO2001078430A1 (fr) 2000-04-11 2000-04-11 Application de rtp et de rtcp dans le transport amr sur reseaux voix sur ip

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