JP2009147825A - Rtp packet transmitter and rtp packet receiver - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a RTP packet transmitter and a RTP packet receiver for transmitting and receiving voice data and application data by a highly universal method with no fear of noise or security problem. <P>SOLUTION: As for the RTP packet which contains application data in payload, a time stamp value applying unit 13 of an IP telephone set 10 on the transmitting side gives the same time stamp value as RTP packet, which was formed last time, to a RTP header of the RTP packet. Even if it is overlapped with the last time stamp value, the sequence number is exact, so the RTP packet is not discarded by an intermediate IP telephone router or a proxy server. On the other hand, the RTP packet in which the application data is included is discarded after the application data is extracted on the receive side, so problem of noise or security does not occur. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to an RTP packet transmitter and an RTP packet receiver, and more particularly to an RTP packet transmitter and an RTP packet receiver for transmitting and receiving voice data and application data.

  As a protocol for transferring video and audio data in real time, RTP (Real-time Transport Protocol) which is a protocol defined by IETF RFC1889 is used. For example, Patent Document 1 discloses a technique for transferring video and audio data by RTP in a system that requires real-time reproduction such as video on demand or remote video conference.

In the system disclosed in Patent Document 1, the RTP packet generation unit executes processing for generating a packet using encoded data as a payload. The RTP packet generator adds an RTP header to the payload data and packetizes it. The RTP header includes version number (v), padding (P), presence / absence of extension header (X), number of transmission sources (Counter), marker information (markerbit), payload type (Payload type), sequence number, time stamp, The fields of synchronization source (source) identifier (SSRC) and contributing source (source) identifier (CSRC) are provided. On the data receiving side, the processing time is controlled when the RTP packet is expanded based on the time stamp added to the RTP header, thereby enabling real-time image or audio reproduction control.
JP 2003-179580 A

  By the way, in an IP telephone using RTP, there are an inband system and an outband system as a system for transferring application data such as other telephone numbers during transfer of voice data. In the inband system, DTMF (Dual-Tone Multi- Frequency) and outband systems include an SIP (Session Initiation Protocol) INFO method. However, since DTMF is an audible sound, there are problems such as noise and security problems and slow communication performance. In addition, the SIP INFO method is not supported by some ISPs (Internet Service Providers) and may not be used.

  The present invention has been made in view of such circumstances, and an object of the present invention is to transmit and receive audio data and application data by a highly versatile method without causing noise or security problems. An RTP packet transmitting device and an RTP packet receiving device that can be used are provided.

  The present invention includes a first RTP packet generating unit that generates a first RTP packet including voice data in a payload, a second RTP packet generating unit that generates a second RTP packet including application data in a payload, RTP packet transmitting means for transmitting the first RTP packet and the second RTP packet generated by the first RTP packet generating means and the second RTP packet generating means to the network, respectively, and the first RTP packet The generation unit assigns a time stamp value to be sequentially added to the RTP header of the first RTP packet, and the second RTP packet generation unit transmits the previous transmission to the RTP header of the second RTP packet. It is an RTP packet transmission device that gives the same time stamp value as the first RTP packet.

  In addition, the present invention provides a first RTP packet generation unit that generates a first RTP packet including voice data in a payload, and a second RTP packet generation unit that generates a second RTP packet including application data in a payload. And RTP packet transmitting means for transmitting the first RTP packet and the second RTP packet generated by the first RTP packet generating means and the second RTP packet generating means to the network, respectively, The RTP packet generation means assigns time stamp values that are sequentially added to the RTP header of the first RTP packet, and the second RTP packet generation means applies to the RTP header of the second RTP packet. A time stamp value smaller than the time stamp value of the first RTP packet transmitted last time is given, A TP packet transmitting apparatus.

  Furthermore, the present invention provides a first RTP packet generating means for generating a first RTP packet including voice data in a payload, and a second RTP packet generating means for generating a second RTP packet including application data in a payload. And RTP packet transmitting means for transmitting the first RTP packet and the second RTP packet generated by the first RTP packet generating means and the second RTP packet generating means to the network, respectively, The RTP packet generation means assigns a time stamp value in which two or more values are sequentially added to the RTP header of the first RTP packet, and the second RTP packet generation means outputs the RTP packet of the second RTP packet. A time stamp that is one greater than the time stamp value of the first RTP packet transmitted last time with respect to the header Imparting value, a RTP packet transmitting apparatus.

  On the other hand, the present invention provides an RTP packet receiving means for receiving RTP packets from a network, an audio data processing means for processing voice data, an application data processing means for processing application data, and an RTP packet received by the RTP packet receiving means. When the time stamp value of the RTP header is larger than the time stamp value of the previously received RTP packet, the data included in the payload of the RTP packet is passed to the voice data processing means, and the RTP packet receiving means receives the RTP RTP packet analysis means for passing data contained in the payload of the RTP packet to the application data processing means when the time stamp value of the RTP header of the packet is the same as the time stamp value of the previously received RTP packet. That is a RTP packet receiver.

  The present invention also provides an RTP packet receiving means for receiving an RTP packet from a network, an audio data processing means for processing voice data, an application data processing means for processing application data, and an RTP packet received by the RTP packet receiving means. When the time stamp value of the RTP header is equal to or greater than the time stamp value of the previously received RTP packet, the data included in the payload of the RTP packet is passed to the voice data processing means, and the RTP packet receiving means receives the RTP RTP packet analysis means for passing data contained in the payload of the RTP packet to the application data processing means when the time stamp value of the RTP packet of the packet is smaller than the time stamp value of the RTP packet received last time Obtain a RTP packet receiver.

  Further, the present invention provides an RTP packet receiving means for receiving an RTP packet from a network, an audio data processing means for processing voice data, an application data processing means for processing application data, and an RTP packet received by the RTP packet receiving means. When the time stamp value of the RTP header of the RTP packet is 2 or more larger than the time stamp value of the RTP packet received last time, the data included in the payload of the RTP packet is passed to the voice data processing means, and the RTP received by the RTP packet receiving means An RTP packet receiving apparatus comprising: an RTP packet analyzing unit that passes data included in a payload of the RTP packet to an application data processing unit when the time stamp value of the packet is larger by one.

  According to the RTP packet transmitting device and the RTP packet receiving device of the present invention, it is possible to transmit and receive voice data and application data by a highly versatile method without causing noise and security problems. .

  Hereinafter, an RTP packet transmitting apparatus and an RTP packet receiving apparatus according to embodiments of the present invention will be described with reference to the accompanying drawings.

  FIG. 1 is a diagram illustrating a configuration example of a communication system in the IP telephone according to the first embodiment. In this embodiment, the RTP packet transmitter and RTP packet receiver according to the present invention are configured as a communication system in an IP telephone.

  As shown in FIG. 1, IP telephones 10 and 20 of this embodiment are implemented as personal computers, and are connected to an IP telephone network 40 via IP telephone routers 30 and 60, respectively. In the IP telephone network 40, there are a redirect server (not shown) in addition to the SIP / proxy server 50 for each medium.

FIG. 2 is a diagram showing a connection by the end-to-end scheme in the IP telephone according to the first embodiment. In the following description, a standard configuration of an IP telephone is assumed. That is, the signaling protocol is SIP (Session Initiation Protocol), the voice data transfer uses RTP, the voice data control protocol uses RTCP (RTP Control Protocol), and the voice CODEC uses G.711 (Pulse code modulation ( PCM) of voice frequencies), each standard is as follows.
SIP ... RFC 3261 etc. RTP ... RFC 3550 etc. RTCP ... RFC 3550 etc. G. 711 ... ITU-T Recommendation

  As shown in FIG. 2, the IP telephones 10 and 20 of this embodiment use a SIP connection a, a voice RTP connection b, and a voice RTCP connection c. The SIP connection a is connected to one or more SIP servers existing in the communication system of the IP telephone, but the voice RTP connection b and the voice RTCP connection c are transmitted between the IP telephones 10 and 20 via a proxy server for each medium. Communication is performed by an end-to-end system, which is a connection system for direct communication.

  FIG. 3 is a functional block diagram showing a software configuration of the transmission-side IP telephone 10 according to the first embodiment. In the following description, for the sake of convenience, the IP telephone on the transmission side is referred to as IP telephone 10 and the IP telephone on the reception side is referred to as IP telephone 20.

  As shown in FIG. 3, the transmission side IP telephone 10 includes a transmission buffer 11, an RTP packet generation unit (first RTP packet generation unit, second RTP packet generation unit) 12, and an RTP port (RTP packet transmission unit). 15 is provided. The transmission buffer 11 is for buffering voice data of voice calls of the IP telephone 10 and application data such as other telephone numbers.

  The RTP packet generator 12 is for sequentially generating RTP packets that include either audio data or application data in the payload. The RTP packet generation unit 12 includes a time stamp value adding unit 13 and a marker bit value adding unit 14.

  The time stamp value giving unit 13 is for giving a time stamp value to the RTP header of each RTP packet. The marker bit value assigning unit 14 is for assigning a marker bit value indicating whether or not application data is included in the payload of the RTP packet to the RTP header of the RTP packet.

  The RTP port 15 is for sequentially transmitting the RTP packets generated by the RTP packet generator 12 to the IP telephone network 40.

  FIG. 4 is a functional block diagram showing a software configuration of the IP telephone 20 on the receiving side according to the first embodiment. As shown in FIG. 4, the receiving side IP telephone 20 includes an RTP port (RTP packet receiving means) 21, an RTP packet correction processing section, a reception buffer 23, and an RTP packet analysis section (RTP packet analysis means, voice data processing means). Application data processing means) 24.

  The RTP port 21 is used for sequentially receiving RTP packets transmitted from the IP telephone 10 from the IP telephone network 40. The RTP packet correction processing unit 22 is for performing correction processing such as missing RTP packets. The reception buffer 23 is for buffering the RTP packet that has been corrected by the RTP packet correction processing unit 22.

  The RTP packet analysis unit 24 sequentially analyzes received RTP packets, sequentially extracts either voice data or application from the RTP packet, and processes the voice data and application. The RTP packet analysis unit 24 includes a time stamp value analysis unit 25 and a marker bit value analysis unit 26.

  The time stamp value analysis unit 25 is for analyzing the time stamp value included in the RTP header of each RTP packet. The marker bit value analysis unit 26 is for analyzing the marker bit value included in the RTP header of each RTP packet.

  FIG. 5 is a diagram illustrating the structure of the RTP header. In FIG. 5, “V = 2” represents a version number (2 bits), and normally 2 is entered. “P” represents padding (1 bit). When 1 is inserted, it indicates that the payload is padded at the end of the packet and is longer than the original length. “X” is a flag set when an extension header is provided immediately after the RTP header (1 bit). “CC” is a CSRC (Contributing Source) count (4 bits) and contains the number of subsequent CSRC identifiers.

  “M” is a marker bit (1 bit). When audio data is included in the payload of the RTP packet, 0 is entered (first value), and application data is included in the payload of the RTP packet. In this case, 1 is entered (second value).

  “PT” is a payload type (7 bits) and represents the type of the subsequent payload. “Sequence number” is a sequence number (sequence number) (16 bits), which is a serial number of an RTP packet that increases by one. A random value is used as the initial value to ensure security.

  “Timestamp” is a time stamp value (32 bits) and represents the time when the oldest data included in the packet occurs. The time stamp value increases at a constant cycle that differs for each medium. The clock frequency that determines the period of the time stamp value is defined for each payload type (PT), and in the case of an audio codec, it is the sampling frequency. For example, 8 kHz sampling G.P. When transmitting a 711 voice codec with a packet length of 20 msec, one RTP packet contains 160 samples of voice data, and the time stamp value is also increased by 160 for each RTP packet. In this way, since the transmitting side reliably counts up by 160, it is possible to predict the next time stamp value on the receiving side. By using this characteristic, application data is included in the payload of the RTP packet. It becomes possible to guess whether or not.

  “Synchrozination Source (SSRC) identifier” is an identifier of a participant in the RTP session (32 bits). The SSRC is a temporary identifier that changes for each session, and is associated with a CNAME (Canonical Name) by RTCP. “Contributing Source (CSRC) identifier” is an identifier of a transmission source prepared for each stream element in the message (32 bits). Each value is the same as each SSRC.

  FIG. 6 is a sequence diagram showing transfer of an RTP packet (first RTP packet) including audio data according to the end-to-end scheme according to the first embodiment. In the example of FIG. 6, the application data is not transferred, the sequence number is incremented by one, and the time stamp value given by the time stamp assigning unit 13 of the IP telephone 10 on the transmission side is the reference clock frequency 8000. Since it is sometimes 160 counts, it shows that it is transferred at intervals of 20 ms. G. is a standard voice CODEC system for IP telephones 10 and 20. In 711, such RTP packet transfer is generally performed. The time stamp value analysis unit 25 of the IP telephone 20 on the receiving side includes a time stamp value obtained by adding 160 counts to the time stamp value of the RTP packet compared to the time stamp value of the RTP packet analyzed last time. When this is analyzed, the RTP packet analysis unit 24 of the IP telephone 20 on the receiving side extracts voice data from the RTP packet.

  FIG. 7 is a sequence diagram illustrating transfer of an RTP packet including voice data and an RTP packet (second RTP packet) including application data according to the end-to-end scheme according to the first embodiment. In this example, application data is transferred together with audio data. In FIG. 7, RTP packets (seq = α + 4, time = v + 480), RTP packets (seq = α + 6, time = v + 640), RTP packets (seq = α + 8, time = v + 800) are RTP packets including application data. It is.

  For RTP packets including application data in these payloads, the time stamp value assigning unit 13 of the IP telephone 10 on the transmission side uses the RTP packet generated last time for the RTP header of the RTP packet including application data in the payload. Give the same timestamp value.

  On the other hand, when the time stamp value analysis unit 25 of the IP telephone 20 on the receiving side analyzes that the time stamp value of the RTP packet includes the same time stamp value as the time stamp value of the previously analyzed RTP packet The RTP packet analysis unit 24 of the IP telephone 20 on the receiving side extracts application data from the RTP packet and discards the packet.

  That is, in this embodiment, the time stamp value of the RTP packet including the application data in the payload overlaps with the time stamp value of the previous RTP packet, but no matter what IP telephone router or media proxy server exists in the middle, Since the sequence numbers of these RTP packets are accurate, the RTP packets are never discarded because it is illegal to have duplicate time stamp values. The RTP packet including the voice data is transferred from the transmission side to the reception side at a period indicated by the time stamp value and used for voice call. On the other hand, an RTP packet including application data is discarded after the application data is extracted on the receiving side, so that noise does not interfere with voice communication using voice data or cause a security problem. . Therefore, in the present embodiment, noise and security problems do not occur, and audio data and application data can be transmitted and received by a highly versatile method.

  FIG. 8 is a diagram showing a connection by the link-by-link method in the IP telephone according to the first embodiment. In this example, the SIP connection a is connected to one or more SIP servers existing in the communication system of the IP telephone, and the voice RTP connection b and the voice RTCP connection c are a proxy server for each medium between the IP phones 10 and 20. Communication is performed by a Link-by-Link method, which is a connection method for performing communication as a gateway.

  FIG. 9 is a sequence diagram showing transfer of an RTP packet including voice data by the link-to-link method according to the first embodiment. In the example of FIG. 9, the application data is not transferred and the sequence number is incremented by one, but the time stamp value given by the time stamp assigning unit 13 of the IP telephone 10 on the transmission side is the reference clock frequency 8000. Since it is sometimes 160 counts, it shows that it is transferred at intervals of 20 ms. Protocol processing by RTP is performed by the IP telephone routers 30 and 60, the proxy server, etc., but the difference from the initial value of the sequence number and the difference from the initial value of the time stamp value do not change. G. is a standard voice CODEC system for IP telephones 10 and 20. In 711, such RTP packet transfer is generally performed. The time stamp value analysis unit 25 of the IP telephone 20 on the receiving side includes a time stamp value obtained by adding 160 counts to the time stamp value of the RTP packet compared to the time stamp value of the RTP packet analyzed last time. When this is analyzed, the RTP packet analysis unit 24 of the IP telephone 20 on the receiving side extracts voice data from the RTP packet.

  FIG. 10 is a sequence diagram illustrating transfer of an RTP packet including voice data and an RTP packet including application data according to the link-to-link method according to the first embodiment. In this example, application data is transferred together with audio data. In FIG. 7, RTP packets (seq = α + 4, time = v + 480), RTP packets (seq = α + 6, time = v + 640), RTP packets (seq = α + 8, time = v + 800) are RTP packets including application data. It is.

  For RTP packets including application data in these payloads, the time stamp value assigning unit 13 of the IP telephone 10 on the transmission side uses the RTP packet generated last time for the RTP header of the RTP packet including application data in the payload. Give the same timestamp value.

  On the other hand, when the time stamp value analysis unit 25 of the IP telephone 20 on the receiving side analyzes that the time stamp value of the RTP packet includes the same time stamp value as the time stamp value of the previously analyzed RTP packet The RTP packet analysis unit 24 of the IP telephone 20 on the receiving side extracts application data from the RTP packet and discards the packet.

  As described above, in the present embodiment, noise and security problems do not occur even in communication using the link-to-link method, and voice data and application data are transmitted and received by a highly versatile method. Is possible.

  In this embodiment, the marker bit value of the RTP header can be used to indicate that application data is included in the RTP payload. That is, the marker bit value assigning unit 14 of the IP telephone 10 on the transmission side sets a marker bit value M = 1 indicating whether or not application data is included in the payload of the RTP packet with respect to the RTP header of the RTP packet. Give. On the other hand, when the marker bit value analysis unit 26 of the IP telephone 20 on the receiving side analyzes that the marker bit value of the RTP packet is M = 1 indicating that application data is included in the payload of the RTP packet. The RTP packet analysis unit 24 of the receiving side IP telephone 20 extracts application data from the RTP packet.

  In the present embodiment, by using the marker bit value as described above, it is possible to more reliably determine whether application data is included in the payload of the RTP packet.

  Hereinafter, a second embodiment of the present invention will be described. FIG. 11 is a sequence diagram illustrating transfer of an RTP packet including voice data and an RTP packet including application data according to the end-to-end scheme according to the second embodiment. In the present embodiment, in FIG. 11, RTP packets (seq = α + 4, time = v + 479), RTP packets (seq = α + 6, time = v + 639), RTP packets (seq = α + 8, time = v + 799), application data It is an included RTP packet.

  For RTP packets that include application data in these payloads, the time stamp value assigning unit 13 of the IP telephone 10 on the transmission side uses the RTP packet generated last time for the RTP header of the RTP packet that includes application data in the payload. Also, a time stamp value less by 1 is assigned.

  On the other hand, the time stamp value analysis unit 25 of the IP telephone 20 on the receiving side indicates that the time stamp value of the RTP packet includes a time stamp value that is one less than the time stamp value of the RTP packet analyzed last time. When the analysis is performed, the RTP packet analysis unit 24 of the IP telephone 20 on the receiving side extracts application data from the RTP packet and discards the packet.

  Also in the communication by the Link-to-Link system shown in FIG. 12, it is possible to transmit / receive RTP packets including application data in the payload in the same manner as described above.

  In this embodiment, the time stamp value of the RTP packet including the application data in the payload is one less than the time stamp value of the previous RTP packet. However, even if any IP telephone router or media proxy server exists on the way, Since the sequence number of the RTP packet is accurate, the RTP packet is never discarded because it is illegal that the time stamp value is 1 less. On the other hand, since the RTP packet including application data has a time stamp value that is one less than the time stamp value of the previous RTP packet, it is reliably discarded after the application data is extracted on the receiving side. Thus, it is possible to more reliably prevent troubles in voice calls using voice data and occurrence of security problems.

  Hereinafter, a third embodiment of the present invention will be described. FIG. 13 is a sequence diagram showing transfer of an RTP packet including voice data and an RTP packet including application data according to the end-to-end scheme according to the second embodiment. In this embodiment, in FIG. 13, RTP packets (seq = α + 4, time = v + 481), RTP packets (seq = α + 6, time = v + 641), RTP packets (seq = α + 8, time = v + 801) are stored in the application data. It is an included RTP packet.

  For RTP packets that include application data in these payloads, the time stamp value assigning unit 13 of the IP telephone 10 on the transmission side uses the RTP packet generated last time for the RTP header of the RTP packet that includes application data in the payload. Also, a time stamp value that is 1 more is added.

  On the other hand, the time stamp value analysis unit 25 of the IP telephone 20 on the receiving side indicates that the time stamp value of the RTP packet includes a time stamp value that is one more than the time stamp value of the RTP packet analyzed last time. When the analysis is performed, the RTP packet analysis unit 24 of the IP telephone 20 on the receiving side extracts application data from the RTP packet and discards the packet.

  In communication using the link-to-link method shown in FIG. 14, RTP packets including application data in the payload can be transmitted and received in the same manner as described above. In this embodiment, the time stamp value of the RTP packet including the application data in the payload is one more than the time stamp value of the previous RTP packet. However, any IP telephone router or media proxy server may be present in the middle. Since the sequence number of the RTP packet is accurate, the RTP packet is not discarded because it is illegal that the time stamp value is 1 larger. On the other hand, since the RTP packet including application data has a time stamp value one more than the time stamp value of the previous RTP packet, the RTP implementation method on the receiving side does not discard the RTP packet having the smaller time stamp value. However, the application data is extracted and discarded on the receiving side, so that the voice call using the voice data is not hindered by the noise and the security problem does not occur. Therefore, in this embodiment, according to the RTP mounting method in the communication system, there is no noise or security problem, and voice data and application data can be transmitted and received by a highly versatile method. It becomes possible.

  Although the embodiment of the present invention has been described above, the present invention is not limited to the above embodiment, and various modifications can be made.

  According to the present invention, during a call by the IP telephones 10 and 20, data exchange between the software of the IP telephones 10 and 20 is possible using the payload of the RTP packet. Thereby, the configuration information held by the software of the IP telephones 10 and 20 and the configuration information held by the devices of the IP telephones 10 and 20 can be exchanged. If such configuration information can be exchanged, new communication (such as IPv6 protocol communication) between the software of the IP telephones 10 and 20 becomes possible.

  That is, in the present invention, for example, in an NGN (Next Generation Network) 80 including an IP telephone network as shown in FIG. 15, the IP telephones 10 and 20 are connected to each other and connected to the IP telephone routers 30 and 60, respectively. It is conceivable to shift to the construction of a new private net including those devices by transmitting the addresses and properties of the liquid crystal television 70 and the digital camera 90 which are different devices.

It is a figure which shows the structural example of the communication system in the IP telephone concerning 1st Embodiment. It is a figure which shows the connection by the End-to-End system in the IP telephone apparatus which concerns on 1st Embodiment. FIG. 3 is a functional block diagram showing a software configuration of a transmission-side IP phone according to the first embodiment. FIG. 3 is a functional block diagram showing a configuration on software in the IP telephone on the receiving side according to the first embodiment. It is a figure which shows the structure of a RTP header. It is a sequence diagram which shows transfer of the RTP packet containing the audio | speech data by the End-to-End system based on 1st Embodiment. FIG. 6 is a sequence diagram showing transfer of an RTP packet including audio data and an RTP packet including application data according to an end-to-end scheme according to the first embodiment. It is a figure which shows the connection by the Link-to-Link system in the IP telephone concerning 1st Embodiment. It is a sequence diagram which shows transfer of the RTP packet containing the audio | voice data by the Link-to-Link system which concerns on 1st Embodiment. FIG. 5 is a sequence diagram showing transfer of an RTP packet including audio data and an RTP packet including application data according to the link-to-link method according to the first embodiment. FIG. 10 is a sequence diagram showing transfer of an RTP packet including audio data and an RTP packet including application data according to an end-to-end scheme according to the second embodiment. FIG. 10 is a sequence diagram illustrating transfer of an RTP packet including audio data and an RTP packet including application data according to the Link-to-Link method according to the second embodiment. FIG. 10 is a sequence diagram illustrating transfer of an RTP packet including audio data and an RTP packet including application data according to an end-to-end scheme according to the third embodiment. FIG. 10 is a sequence diagram illustrating transfer of an RTP packet including audio data and an RTP packet including application data according to the link-to-link method according to the third embodiment. It is a figure which shows the example which connected another device to the communication system by an IP telephone.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... IP telephone, 11 ... Transmission buffer, 12 ... RTP packet generation part, 13 ... Time stamp value provision part, 14 ... Marker bit value provision part, 15 ... RTP port, 20 ... IP telephone, 21 ... RTP port, 22 ... RTP packet correction processing unit, 23 ... reception buffer, 24 ... RTP packet analysis unit, 25 ... time stamp value analysis unit, 26 ... marker bit value analysis unit, 30 ... IP telephone router, 40 ... IP telephone network, 50 ... SIP / proxy server, 60 ... IP telephone network, 70 ... liquid crystal television, 80 ... NGN, 90 ... digital camera.

Claims (8)

First RTP packet generation means for generating a first RTP packet including voice data in a payload;
Second RTP packet generation means for generating a second RTP packet including application data in a payload;
RTP packet transmitting means for transmitting the first RTP packet and the second RTP packet generated by the first RTP packet generating means and the second RTP packet generating means, respectively, to the network;
With
The first RTP packet generation means gives a time stamp value sequentially added to the RTP header of the first RTP packet,
The second RTP packet generation means gives the same time stamp value as the first RTP packet transmitted last time to the RTP header of the second RTP packet.
RTP packet transmitter. First RTP packet generation means for generating a first RTP packet including voice data in a payload;
Second RTP packet generation means for generating a second RTP packet including application data in a payload;
RTP packet transmitting means for transmitting the first RTP packet and the second RTP packet generated by the first RTP packet generating means and the second RTP packet generating means, respectively, to the network;
With
The first RTP packet generation means gives a time stamp value sequentially added to the RTP header of the first RTP packet,
The second RTP packet generation means assigns a time stamp value smaller than the time stamp value of the first RTP packet transmitted last time to the RTP header of the second RTP packet. . First RTP packet generation means for generating a first RTP packet including voice data in a payload;
Second RTP packet generation means for generating a second RTP packet including application data in a payload;
RTP packet transmitting means for transmitting the first RTP packet and the second RTP packet generated by the first RTP packet generating means and the second RTP packet generating means, respectively, to the network;
With
The first RTP packet generation means assigns a time stamp value in which two or more values are sequentially added to the RTP header of the first RTP packet;
The second RTP packet generation means assigns a time stamp value larger than the time stamp value of the first RTP packet transmitted last time to the RTP header of the second RTP packet. apparatus. The first RTP packet generation means assigns a first marker bit value to the RTP header of the first RTP packet,
4. The RTP packet transmission device according to claim 1, wherein the second RTP packet generation unit assigns a second marker bit value to an RTP header of the second RTP packet. 5. . RTP packet receiving means for receiving RTP packets from the network;
Audio data processing means for processing audio data;
Application data processing means for processing application data;
When the time stamp value of the RTP header of the RTP packet received by the RTP packet receiving means is larger than the time stamp value of the RTP packet received last time, the data included in the payload of the RTP packet is converted to the voice data processing means. When the time stamp value of the RTP header of the RTP packet received by the RTP packet receiving means is the same as the time stamp value of the RTP packet received last time, the data included in the payload of the RTP packet is transferred to the application data. An RTP packet receiving device comprising: RTP packet analysis means for passing to processing means. RTP packet receiving means for receiving RTP packets from the network;
Audio data processing means for processing audio data;
Application data processing means for processing application data;
When the time stamp value of the RTP header of the RTP packet received by the RTP packet receiving means is greater than or equal to the time stamp value of the previously received RTP packet, the data included in the payload of the RTP packet is converted to the voice data processing means. When the time stamp value of the RTP packet of the RTP packet received by the RTP packet receiving means is smaller than the time stamp value of the RTP packet received last time, the data included in the payload of the RTP packet is processed by the application data processing RTP packet receiving device comprising: RTP packet analyzing means for passing to the means. RTP packet receiving means for receiving RTP packets from the network;
Audio data processing means for processing audio data;
Application data processing means for processing application data;
When the time stamp value of the RTP header of the RTP packet received by the RTP packet receiving means is two or more larger than the time stamp value of the RTP packet received last time, the data included in the payload of the RTP packet is converted to the voice data processing means. RTP packet analyzing means for passing data included in the payload of the RTP packet to the application data processing means when the RTP packet receiving means is larger by one than the time stamp value of the RTP packet received by the RTP packet receiving means. Packet receiver.   The RTP packet analyzing means, when the marker bit value included in the RTP header of the RTP packet received by the RTP packet receiving means is a first value, the data included in the payload of the RTP packet as the voice data processing means And when the marker bit value included in the RTP header of the RTP packet received by the RTP packet receiving means is a second value, the data included in the payload of the RTP packet is passed to the application data processing means. Item 4. The RTP packet receiver according to any one of Items 1 to 3.

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