EP1433301A1

EP1433301A1 - Method and device for signalling a codec negotiation over heterogeneous signalling networks

Info

Publication number: EP1433301A1
Application number: EP02774332A
Authority: EP
Inventors: Helmut Schmidt; Bernd Siegwart; Erik Margraf
Original assignee: Siemens AG; Nokia Siemens Networks GmbH and Co KG
Current assignee: Nokia Solutions and Networks GmbH and Co KG
Priority date: 2001-10-05
Filing date: 2002-09-11
Publication date: 2004-06-30
Also published as: WO2003032615A1; US20040258016A1

Abstract

The invention relates to a method and a device for signalling a codec negotiation over heterogeneous signalling networks. In a heterogeneous network environment, interfaces can be present, for a transmission, which are governed by limitations in terms of a codec negotiation. Especially cases in which only a limited set of codecs is supported and/or signalled in the target network, or in which the target network only has a limited number of elements, for example one element, from which proposed codec lists can be extracted or evaluated, present problems. If this is the case, as a result, connections are rejected if non-supported codecs are signalled. The aim of the invention is to provide an improved method for signalling codec negotiations, whereby as few as possible signalling calls are rejected. To this end, a significant idea of the invention is that the list of supported codecs is resorted before the signalling call in the emission network. The codec list is resorted in such a way that a codec which is (highly likely to be) supported by a terminal in the reception network is placed in first position in the list.

Description

description

Method and arrangement for signaling a codec negotiation over heterogeneous signaling networks

The invention relates to a method for signaling a codec negotiation over heterogeneous signaling networks according to the preamble of claim 1 and an arrangement therefor.

So far, communication connections for the transmission of voice have mainly been established in a connection-oriented manner. For this purpose, only a physical connection is provided for signal transmission between two communication endpoints, which is reserved for the entire time of the connection. This is also referred to as line-oriented transmission, static routing (circuit switching) or circuit switching.

With the advent of packet-oriented data networks (packet

Switching), such as the Internet, is offered in the fixed network area in comparison to connection-oriented telecommunication, communication which is reduced in cost or also improved possibilities for integrating value-added services. This is due in particular to the efficient capacity utilization of a connection, since in contrast to circuit switching, packet switching does not occupy the physical transmission medium for the entire duration of the connection. For some time now there have been mature business solutions for the transmission of voice via packet-based

Protocols, such as the frame relay, are available. The idea of also packing voice into IP packets was not too distant. VoIP (Voice over IP) solutions are particularly interesting for companies that have an IP router network and use the public network to make calls.

All VoIP solutions work largely the same Scheme. The data to be transported are divided into individual data packets, each data packet receiving an address code which identifies the recipient of the transmission. The individual data packets are then transmitted independently of one another - they can even use different transmission paths. The principle of packet switching is defined in various standards, a well-known standard is described in ITU-T recommendation x.25.

VoIP will be essential for the future

Predicted voice communication. Here, the language is digitized and, if necessary, compressed by hardware or software (source-coded), the compressed language then representing the useful data area of the IP packets. Dialed numbers are converted into IP addresses that are sent to the IP header as destination information. The IP packets are now transported to the remote site of the voice connection via several network nodes distributed in the data network. The remote station saves the incoming packets and reassembles them in the correct order. If a packet is damaged or lost, it will not be sent again. At the remote station, the voice information is extracted from the packets, which is then fed to a coding device, in which the information is then inversely source and / or channel coded and then made audible via suitable hardware.

Hardware and / or software modules which combine the functions of a coder and a decoder are referred to as codecs, since the transmission of information between two points often involves transmission in both directions. Sometimes the codec is tailored to the characteristics of an input signal, such as voice and / or video signals. The practical implementation takes place either as hardware by DSPs (Digital Signal Processors) or by codec programs implemented in software. In order to minimize the storage space required for a complex data stream, for example audio and / or video data, the data are compressed according to defined algorithms. A decompression algorithm is required to use the data, which cancels the compression. This means that every compression has a corresponding decompression that inverts exactly this compression. The hardware and software solutions created for this are usually also referred to as codecs. A data stream encoded or compressed with a certain codec can only be decoded or decompressed with this codec.

Known codecs are, for example, G.711, G.722, G.723, G.726, G.728, G.729 or GSM codecs for the mobile radio sector. G.711 is a recommendation of the International Telecommunication Union (ITU), which describes the digitization of audio data in telephone quality (3.1 kHz bandwidth) with a data rate of 56 kbit / s or 64 kbit / s. The process is called pulse code modulation and is used in the analog telephone network or in ISDN. G.723 defines a voice compression process for use in narrowband, multimedia applications with data rates of 6.4 and 5.3 kbit / s. The method was originally developed by the ITU with a view to its use in narrowband video conference systems and is now increasingly used in IP telephony. Other codecs include MP3 (MPEG layer III audio) for high-quality music data on the Internet, H.261 or H.263 for video conferences with low or medium quality or Sorenson video for high-quality WWW video data.

These codecs are used to encode the data to reduce storage space requirements or to speed up data transmission. As already mentioned above, the codec used for sending the data for decoding / decompressing the received data must be available on the receiver side. From this point of view, To ensure ten transmission, the sending terminal creates a codec list and brings about agreement with the received terminal by means of a so-called negotiation regarding the codec to be used for sending and receiving. The codec negotiation takes place in connection with the call signaling. During the codec negotiation, the received terminal device selects a codec it supports from the received codec list. This selection is signaled back to the sending terminal.

In a heterogeneous network environment, there may be interfaces during transmission that are subject to restrictions with regard to codec negotiation. In particular, the cases that only a limited set of codecs are supported and / or signaled in the target network or that a target network can only extract or evaluate a limited number of elements, for example one element, from the proposed codec list. As a result, connections are rejected if unsupported codecs are signaled.

It is therefore an object of the present invention to provide an improved method for signaling codec negotiations in which as few signaling calls as possible are rejected.

This object is achieved by a method which is the subject of claim 1. An essential idea of the invention is that the codec list of the supported codecs is re-sorted before the signaling call in the transmission network. The codec list is rearranged in such a way that a codec is placed at the top of the list, which is (most likely) supported by a terminal in the receiving network. The list is sorted according to a defined and administratively specified order. It makes sense to put a codec in the first place in the defined order, which is supported by end devices in as many networks as possible. This If the number of signaling calls is significantly lower, the terminal rejects the call in the receiving network and thus increases the success rate of the transmissions. The term terminal is understood above and hereinafter to mean a terminal or a switching center. It is essential that a corresponding call signaling protocol is terminated there.

It is advantageous to use the method according to the invention for transmissions, for example for tariff reasons, in which a transit network is interposed between the transmission network and the reception network. This is e.g. B. useful when using the method in IP telephony. For example, there can be two end devices in the public telephone network, which thus serves both as a transmitting and a receiving network. An IP network, for example the public Internet, is used as the transit network. This enables, for example, a telephone call between Munich and Hamburg to be made at a lower cost than a comparable long-distance call made entirely over the public telephone network.

A terminal of the transit network preferably signals the codec list unchanged to a terminal of the receiving network. The terminal in the transit network likewise preferably selects a limited number of elements, in particular only the first element, from the codec list. Even if the terminal of the transit network only supports the first element of the codec list, all information, in particular the complete codec list of the transmitting network, is still forwarded to the receiving network. The codec negotiation between the terminal in the receiving network and the terminal in the transmitting network is not subject to any restrictions as a result of a possibly limited support for the codec list of the terminal in the transit network.

An arrangement according to the invention has a transmission network Terminal (terminal device in the narrower sense or an interworking point (IWP)) that supports at least one codec, and a receiving network with a terminal that also supports at least one codec. The arrangement also has a sorting device for sorting the codec list.

An advantageous embodiment of the arrangement according to the invention has a sorting device which comprises a database for storing a defined sequence of codecs in the codec list. Furthermore, the sorting device has a processor for executing the sorting of the codec list.

In a preferred arrangement of the invention, the terminal in the transmission network corresponds to the H.323 standard. The H.323

Standard is a recommendation of the International Telecommunication Union (ITU) and describes the transport of multimedia data over IP-based networks, in particular the transport with bidirectional real-time communication connections.

Further advantageous embodiments result from the subclaims and the following description of the preferred exemplary embodiments, which are explained in more detail with reference to the figures. Here show:

1 shows a structure of a communication terminal according to the H.323 standard,

2 shows a simplified representation of a network environment with two networks,

3 is a simplified representation of a network environment with two networks for a method of the present invention and

Fig. 4 is a simplified representation of a network environment with three networks for a method according to the present Invention.

1 outlines a system structure of a terminal 10 according to the H.323 standard, which is a recommendation of the International Telecommunication Union (ITU) and was developed for video conferences via LANs (Local Area Networks) and WANs (Wide Area Networks) , The H.323 standard takes into account data transmission characteristics in LANs and other packet-switching networks, such as fluctuating data rates and delays. Overall, H.323 is generally intended for use over networks that do not provide guaranteed QoS (Quality of Service) for the duration of the connection.

H.323 uses the protocols known from the Internet, UDP (User Datagram Protocol) and RTP (Real-Time Protocol). A protocol device 12 defines the coding of audio signals and video signals. There are separate codecs for each data category, audio / video data, data packets or control signals, which are also standardized. Which codec is used for communication depends on the available resources (computer power, transmission bandwidth) and the desired quality and is determined by the system controller 14 when the connection is established. For this purpose, the control system 14 uses standardized codecs, for example G.711, G.722, G.723 and MPEG-1 as audio codecs, and H.261 and H.263 as video codecs. However, the coding of the data with codecs and the code signaling are not bound to the IP as a transport layer. Other transport layers, for example ATM, can also be used.

In the audio codecs 16, the G.723 standard for VoIP transmission is of particular importance, since a data stream with a data rate of 5.3 kbits / s and good voice quality is available at the output of a coding according to this standard. For the transmission of moving image material via VoIP networks, the video data are coded using the video codecs 18. primed. The H.263 standard is particularly important here, since this codec compresses video data to a data rate of less than 64 kbit / s.

The connection management in a system controller 14 is based on signaling protocols, for example H.245 and the H.225 protocol based on Q.931. Terminal 10 is connected via a LAN interface 8, for example to a gateway. Several H.323 systems can be connected to each other via a LAN network. Terminals in this network can communicate with terminals in other networks via a gateway.

A heterogeneous network environment, in which two networks 22, 26 are connected via a gateway 20, is outlined in FIG. 2 for a signaling method. A first network 22 is shown, which has a terminal 10 according to FIG. 1 and a gatekeeper 24. The second network 26 shown in turn has a terminal 28 according to an ISUP + standard. The ISUP + standard does not really mean a standard. ISUP + in a proprietary extension of the ISUP standard on the migration to BICC.

The gateway 20 transitions from the first network 22 to the second network 26 and the associated conversion between different transmission formats. Gateways are used, for example, to connect the public telephone network to the Internet. The gatekeeper 24 has the task of checking the access authorizations of the users when establishing the connection, performing address conversions and managing the bandwidth available for the communication. The second network can also have a gatekeeper without restricting generality, even if it is not shown in this example.

In the event of a signaling call, the terminal 10 signals the terminal 28 a codec list via the gateway 20, for example wise with Codec G.723 in the first place. A data stream of only 5.3 kbit / s is sufficient for this standard. However, since terminal 28 does not have this G.723 codec, call setup between terminal 10 and terminal 28 is terminated by the latter. Terminal 28 rejects the signal call from terminal 10.

3 shows an arrangement of a heterogeneous network environment consisting of two networks 32, 36, which are connected via the gateway 30, in which the method according to the invention comes into play. The networks 32, 36 each have a terminal or a switching center 10, 38 with respective codecs, which are compiled in a codec list. Terminal 10 is a terminal according to the H.323 standard, whereas terminal 38 is configured according to the ISUP + standard. With the components of FIG. 3 described so far, this arrangement corresponds approximately to the arrangement from FIG. 2. The structure and function of the gatekeeper 34 also corresponds to the structure and function of the gatekeeper 24 and is not described again here.

In contrast to the arrangement from FIG. 2, the network 32 additionally has a sorting device 40. The sorting device 40 manages the codec list of all codecs that the terminal 10 supports. For this purpose, the sorting device 40 has a database 42 for storing the codec list. In addition, the sorting device 40 has a re-sorting stage 44 in order to generate the codec list which the terminal 10 signals to the terminal 38 via the gateway 30. The sorting device 40 does not have to be assigned to the network 32. For example, it can also be assigned to the gateway 30 or generally to an IWP between two networks.

The primary list of the codecs supported by the terminal 10 is rearranged in the sorting device 40 in relation to the sequence stored in the database 42. For example, this list can be rearranged in such a way that position is no longer codec G.723, but codec G.711. Terminal 10 now signals this rearranged list to terminal 38 via gateway 30. Since the terminal 38 also supports the G.711 standard, the signaling call from the terminal 10 is not rejected by the terminal 38, a connection between the two terminals 10, 38 is thus established.

A network arrangement is sketched in FIG. 4, which comprises the elements of the arrangement from FIG. 3 and additionally a network 32 '. The configuration of the networks 32, 36 from FIG. 4 corresponds to that of the networks 32, 36 from FIG. 3 and will not be described again at this point. The additional network 32 'comprises a gatekeeper 34', a terminal 10 'according to the H.323 standard and a sorting device 40'. The sorting device 40 'also has a database 42' for storing the codec list and a re-sorting stage 44 'for sorting the codec list.

Terminal 10 signals terminal 10 'that a call is being set up. However, this does not take place via a direct connection between them, but via the gateway 30, the network 36 and the gateway 30 '. 4, the networks 32 and 32 'are shown as two similar networks. However, it is also possible that the network 32 'is identical to the network 32, or else that the two networks 32 and 32' are different. For example, "networks" 32, 32 'could be different exchange areas of the public telephone network and network 36 could be the Internet.

In the constellation shown in Fig. 4, network 36 is used as a transit network. Terminal 38 does not support all codecs that support terminals 10, 10 '. In the case of an unsorted signaling call from terminal 10, terminal 38 would reject the call if it was first

Instead of the codec list sent, there is a codec that Terminal 38 does not support. Forwarding the signal This would not result in a call from Terminal 10 to Terminal 10 '. The sorting unit 40 therefore sorts the codec list of the codecs supported by the terminal 10 in such a way that a codec which the terminal 38 also supports is in the first position. Therefore, terminal 38 does not reject the signaling call, but instead calls terminal 10 '.

The terminal 10 is then connected to the terminal 10 'via the terminal 38. The codec list re-sorted by the sorting device 40 is fully signaled further from the terminal 38 to the terminal 10 '. Terminal 10 'thus receives a codec list with all codecs supported by terminal 10, even if some of these codecs in this list are not supported by terminal 38. The codec list of the terminal 38, which is restricted in comparison to the codec lists of the terminals 10, 10 ', has no restrictive effect on the signaling of the codec list from the terminal 10 via the gateway 30, the terminal 38 and the gateway 30' Terminal 10 '.

Network environments were shown in FIGS. 3 and 4, in each of which the sending network 32 has a sorting device 40 for sorting the codec list. The network 36 has no corresponding sorting device in both representations. However, it is obvious that in practice the sorting device shown here as an illustration of the invention as a receiving or transit network can also have such a sorting device. It is equally clear that the network 32 'in FIG. 4 does not necessarily require the sorting device 40' in the situation described above to use the method according to the invention.

The implementation of the invention is not limited to the examples described and the aspects highlighted above, but is also possible within the scope of the claims in a large number of modifications which are within the scope of professional action.

Claims

claims

1. A method for signaling between switching nodes or terminals (10, 10 ', 28) in networks, the terminals (10, 10', 28) supporting at least one codec in each network and the method comprising the following steps: establishing a connection between a transmitting terminal (10) of the transmitting network (32) and a receiving terminal (10 ', 38) of a receiving network (32', 36) via an interface (30),

Signaling a codec list of the transmitting terminal (10) to the receiving terminal (10 ', 38), which contains all codecs supported by the transmitting terminal (10), so that the method has further steps:

Sorting the codec list of the codecs supported by the transmitting terminal (10) such that the list has a defined order of the codecs, and negotiating the terminals (10, 10 ', 38) of the transmitting network (32) and the receiving network (32', 36 ) according to the resorted codec list.

2. The method of claim 1, d a d u r c h g e k e n n z e i c h n e t that the step of establishing a connection comprises the following steps:

Establishing a connection between the transmitting terminal (10) of the transmitting network (32) with a terminal (38) of a transit network (36) via a first interface (30), establishing a connection between the terminal (38) of the transit network (36) and the receiving terminal (10 ') of the receiving network (32') via a further interface (30 ').

3. The method according to claim 2, characterized in that the transmission network (32) and the reception network (32 ") are identical.

4. The method according to claim 2 or 3, so that the terminal (38) of the transit network (36) signals the codec list unchanged to the receiving terminal (10 ') of the receiving network (32').

5. The method according to any one of the preceding claims, that the terminal (38) of the transit network (36) evaluates a limited number of elements, in particular only the first element, of the codec list.

6. Arrangement for carrying out the method according to one of claims 1 to 5, comprising: a transmission network (32) with a transmission terminal (10) that supports a plurality of codecs and creates and sends a codec list of the supported codecs, a reception network ( 32 ', 36) with a receiving terminal (10', 38) which supports at least one codec, and an interface (30, 30 ') via which the transmitting and receiving network (32, 32', 36) are connected to one another and which Codec list is transmitted, characterized in that a sorting device (40) for sorting the codec list is provided in the transmission network (32).

7. Arrangement according to claim 6, characterized in that the sorting device (40, 40 ') comprises the following: a database (42, 42') for storing a defined sequence of codecs in the codec list, a sorting stage (44, 44 ') for re-sorting the codec list according to a codec prioritization in the receiving network.

8. Arrangement according to claim 6 or 7, characterized in that the sorting device (40, 40 ') assigned to the transmitting terminal is arranged.

9. Arrangement according to claim 6 or 7, so that the sorting device (40, 40 ') is assigned to a gateway controller of the interface (30, 30').