CN100542132C - voice transmission method based on transmission control protocol - Google Patents
voice transmission method based on transmission control protocol Download PDFInfo
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- CN100542132C CN100542132C CNB200510034465XA CN200510034465A CN100542132C CN 100542132 C CN100542132 C CN 100542132C CN B200510034465X A CNB200510034465X A CN B200510034465XA CN 200510034465 A CN200510034465 A CN 200510034465A CN 100542132 C CN100542132 C CN 100542132C
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Abstract
A kind of voice transmission method based on transmission control protocol, voice based on transmission control protocol are connected, between voice transmitting terminal and receiving terminal, set up at least two transmission control protocols and is connected, and the message that will contain speech data sends to each transmission control protocol connection according to certain regular allocation for each voice channel; Receiving terminal is temporarily stored in the buffering area and resequences after receiving the packet that comes from each transmission control protocol connection, obtains the sequence of message identical with transmitting terminal; The tone decoding system of receiving terminal obtained data and decodes from buffering area according to the predetermined cycle.The present invention has overcome prior art when using Transmission Control Protocol voice-bearer information, can be because packet loss brings the serious deficiency that descends of voice quality, reduce the interrupted probability of voice, improved the reliability and the practicality that connect transferring voice by TCP, improved speech quality.
Description
Technical field
The present invention relates to communication technical field, relate in particular to a kind of voice transmission method based on transmission control protocol (TCP).
Background technology
Along with the rise of NGN (next generation network), a large amount of IP phone machines are used in enterprise or the family at present.The own LAN system of enterprise generally all is to be connected into internet (Internet) by Proxy (agency service) or fire compartment wall.In order to guarantee the safety of intranet, avoid internal enterprise resources to be subjected to outside attack, a large amount of enterprise firewalls has only been opened HTTP (HTML (Hypertext Markup Language)) agency, and other access mode all is under an embargo.Under this situation, in order to allow VoIP (Voice over Internet Protocol, be IP phone) can normally use, generally can use the HTTP tunneling technique, promptly set up a HTTP tunnel between VoIP terminal in local area network (LAN) and the tunnel server on the Internet, this tunnel can pass through proxy, and the voice signaling of VoIP all can be packaged into the HTTP message and transmit in the tunnel.Because HTTP is carried on the TCP (transmission control protocol), so in fact also be with voice bearer on TCP.
At present along with the development of NGN, the application of voip technology more and more widely, voice message transmits by IP network, greatly reduces the investment and the operation cost of communication network.After voice message transmits by packet network, time delay, shake, packet loss that the three big key elements that influence voice quality are network.General VoIP system all has the ability of certain anti-packet loss (Discarded Packets compensation), anti-jitter (Jitter Buffer).The packet loss at random of small probability just reduces the quality of voice, but understands session content and not influence of information interchange for the people, and the speech quality of this moment still can be accepted.Along with the deterioration of network condition, serious packet loss and shake will damage the integrality after voice signal restores, and cause voice quality seriously to descend, and the session both sides be understood session content impact; And time delay allows the both sides of session feel in the session of carrying out the intercom formula greatly when allowing the people that obvious sensation is arranged, and causes people's interchange difficulty.
Generally voice all are to be packaged into UDP (User Datagram Protoco (UDP)) bag, transmit by IP network.UDP does not guarantee the reliability transmitted, but its characteristic that " transmits as possible " can guarantee the real-time of data.In some limited network condition, the UDP message is blocked, and can't freely transmit.Isolate as Intranet and public network, can only rely on the Proxy intercommunication.For the consideration of network security, the proxy of most of enterprise has forbidden udp protocol, has just opened the HTTP visit based on Transmission Control Protocol, and the VoIP that is carried on this moment on the UDP bag can't pass through proxy.For VoIP can be continued to use in this limited network environment, generally adopt the HTTP tunneling technique, be about to voice packet and be packaged into the HTTP message, be carried on the Transmission Control Protocol again and transmit.Though the HTTP tunneling technique can solve the problem that proxy passes through, there is a defective, promptly Network Packet Loss, shake will cause voice quality and seriously influence arbitrarily.
Below the reason of this defective is made a concrete analysis of:
Be illustrated in figure 1 as the existing voice transfer principle schematic that connects based on TCP, Transmission Control Protocol has been realized the order-preserving and the reliable transmission of transmission data strictnesses.In order to realize the order-preserving of data, receiving terminal (as terminal B among Fig. 1) carries out buffered and rearrangement to the message of each reception, and it is consistent with transmitting terminal (as terminal A among Fig. 1) to guarantee to receive data, handles just can forward the data to upper layer module.If the data-bag lost of front, the data of back will be temporarily stored in buffering area and wait for the transmitting terminal retransmission data.Terminal A sends to terminal B with voice message 1,2,3 order in the TCP connection among Fig. 1.
Simultaneously, in order to be implemented in the reliability of transmitting on insecure IP network, Transmission Control Protocol has designed the overtime retransmission mechanism of a cover: receiving terminal (as terminal B among Fig. 1) must be confirmed the message of transmitting terminal (as terminal A among Fig. 1); When transmitting terminal is not received the affirmation message of receiving terminal, transmitting terminal will wait for a period of time; When transmitting terminal is overtime when not receiving the affirmation message of receiving terminal, will retransmit; Retransmission process will continue repeatedly, until transmitting terminal obtains recipient's affirmation, perhaps retransmit number of times and will reach the predetermined upper limit; Overtime duration generally all has about 1S for the first time, has " exponential backoff (exponential backoff) " relation between follow-up overtime duration and the primary overtime duration, and in some system, maximum overtime duration can reach 64 seconds.Simultaneously Transmission Control Protocol has guaranteed the order of message transmission, the inevitable and transmitting terminal of the order of the message that destination applications is received consistent, and not successfully before the transmission, next message will get clogged at last message.
Message dropping that the network reason causes and shake are difficult to avoid.When using the TCP voice-bearer, if the affirmation message dropping that receiving terminal sends, perhaps the repeatedly message dropping of transmitting terminal all will cause transmitting terminal wait and overtime repeating transmission, thereby cause that receiving terminal produces long data interruption, influences voice quality.So use Transmission Control Protocol voice-bearer information under relatively poor network environment, the result that packet loss brings will be intolerable, can understand session content to the session both sides and cause and have a strong impact on.
Summary of the invention
Technical problem to be solved by this invention is: overcome prior art when using Transmission Control Protocol voice-bearer information, can provide a kind of voice transmission method based on transmission control protocol because packet loss brings the serious deficiency that descends of voice quality, improve speech quality.
The present invention solves the problems of the technologies described above the technical scheme that is adopted to be:
This voice transmission method based on transmission control protocol (TCP) may further comprise the steps:
A, to connecting based on the voice of transmission control protocol, between voice transmitting terminal and receiving terminal, set up at least two transmission control protocols and is connected, and the message that will contain speech data sends to each transmission control protocol connection according to certain regular allocation for each voice channel;
B, receiving terminal are temporarily stored in the buffering area and resequence after receiving the packet that comes from each transmission control protocol connection, obtain the sequence of message identical with transmitting terminal;
The tone decoding system of C, receiving terminal obtained data and decodes from buffering area according to the predetermined cycle.
The tone decoding system of receiving terminal is from the buffering area reading of data, and when running into message dropping, the tone decoding system carries out Discarded Packets compensation and handles, and the voice signal of losing is compensated, and recovers voice from incomplete sequence of message.
In the described steps A, between voice transmitting terminal and receiving terminal, be preferably each voice channel and set up three transmission control protocols and be connected.
Beneficial effect of the present invention is: the present invention sets up a plurality of TCP for each voice channel and connects, the message that will contain speech data simultaneously is assigned in turn in each TCP connection and sends, with respect to single TCP connectivity scenario, eliminated the data jamming that packet loss causes, thereby reduced the interrupted probability of voice, improve the reliability and the practicality that connect transferring voice by TCP, improved speech quality.
Description of drawings
Fig. 1 is the existing voice transfer principle schematic that connects based on TCP;
Fig. 2 the present invention is based on the voice transfer principle schematic that TCP connects.
Embodiment
With embodiment the present invention is described in further detail with reference to the accompanying drawings below:
When the Transmission Control Protocol voice-bearer, Network Packet Loss may will cause that voice interrupt for a long time, its basic reason is the order-preserving of Transmission Control Protocol and the blocking-up that reliable transmission has caused being carried on the voice message on the Transmission Control Protocol, but in fact losing of individual voice bag almost can be ignored to the influence of voice messaging.The present invention proposes the speech quality that following scheme is improved the voice messaging that the TCP mode carries for this reason:
Be illustrated in figure 2 as and the present invention is based on the voice transfer principle schematic that TCP connects, set up at least two TCP for each voice channel and is connected (being connected to example with three TCP among the figure describes) between transmitting terminal (terminal A in as figure) and receiving terminal (terminal B in as figure), the message that will contain speech data is simultaneously interleaving according to certain rule (as in turn) and pass through each TCP connection transmission.
There is a VoIP voice conversation between terminal A and the terminal B, between terminal A and terminal B, sets up three TCP and be connected to come transmitting voice information; Transmitting terminal is assigned to voice message 1,2,3,4,5,6......3n+1,3n+2,3n+3 in turn in three TCP connections and sends, connecting 1 as message 1 by TCP sends, message 2 connects 2 by TCP and sends, message 3 connects 3 by TCP and sends, message 4 connects 1 by TCP and sends, and the rest may be inferred, and message 3n+1 connects 1 by TCP and sends, message 3n+2 connects 2 by TCP and sends, and message 3n+3 connects 3 by TCP and sends.
Receiving terminal is temporarily stored in the buffering area and resequences after receiving the packet that comes from three TCP connections, finally obtains the sequence of message identical with transmitting terminal.The tone decoding system of receiving terminal obtains data and decodes from buffering area according to the predetermined period of strictness.
Like this, when packet loss appearred in TCP connection 1, the follow-up data that TCP connects on 1 got clogged, but the data of at this time TCP connection 2, TCP connection 3 are still in normal delivery; After receiving terminal receives that TCP connects 2, TCP connects 3 data, can recover an incomplete sequence, as *, 2,3, *, 5,6 ..., * herein represents that TCP connects 1 message that blocks, this message reality does not exist in buffering area.Follow-up tone decoding system is all the time periodically from the buffering area reading of data, and when running into * (message dropping), the tone decoding system will carry out Discarded Packets compensation and handle, and the voice signal of losing is compensated.Can recover voice from an incomplete sequence of message like this, the quality of these voice can descend to some extent.If TCP connects 1 just short time interruption (as 1～2 second), then the decline of voice quality can be discovered by people's ear hardly.If long-time the interruption, then people's ear can feel obviously that voice quality descends, but can voice interruption not occur as the voice transfer of prior art.
Actual test shows, recovering voice in this case can most clearly be understood by the people, with respect to single TCP connectivity scenario, eliminated the data jamming that packet loss causes, reduced the interrupted probability of voice, improve the reliability and the practicality that connect transferring voice by TCP, improved speech quality.
Those skilled in the art do not break away from essence of the present invention and spirit, can there be the various deformation scheme to realize the present invention, the above only is the preferable feasible embodiment of the present invention, be not so limit to interest field of the present invention, the equivalent structure that all utilizations specification of the present invention and accompanying drawing content are done changes, and all is contained within the interest field of the present invention.
Claims (3)
1, a kind of voice transmission method based on transmission control protocol is characterized in that, may further comprise the steps:
A, to connecting based on the voice of transmission control protocol, between voice transmitting terminal and receiving terminal, set up at least two transmission control protocols and is connected, and the message that will contain speech data sends to each transmission control protocol connection according to certain regular allocation for each voice channel;
B, receiving terminal are temporarily stored in the buffering area and resequence after receiving the packet that comes from each transmission control protocol connection, obtain the sequence of message identical with transmitting terminal;
The tone decoding system of C, receiving terminal obtained data and decodes from buffering area according to the predetermined cycle.
2, the voice transmission method based on transmission control protocol according to claim 1, it is characterized in that: the tone decoding system of receiving terminal is from the buffering area reading of data, when running into message dropping, the tone decoding system carries out Discarded Packets compensation and handles, the voice signal of losing is compensated, from incomplete sequence of message, recover voice.
3, the voice transmission method based on transmission control protocol according to claim 1 and 2 is characterized in that: in the described steps A, set up three transmission control protocols for each voice channel and be connected between voice transmitting terminal and receiving terminal.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB200510034465XA CN100542132C (en) | 2005-04-28 | 2005-04-28 | voice transmission method based on transmission control protocol |
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| Application Number | Priority Date | Filing Date | Title |
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| CNB200510034465XA CN100542132C (en) | 2005-04-28 | 2005-04-28 | voice transmission method based on transmission control protocol |
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| CN1855878A CN1855878A (en) | 2006-11-01 |
| CN100542132C true CN100542132C (en) | 2009-09-16 |
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| CNB200510034465XA Expired - Fee Related CN100542132C (en) | 2005-04-28 | 2005-04-28 | voice transmission method based on transmission control protocol |
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Families Citing this family (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102253987B (en) * | 2011-07-01 | 2012-12-12 | 中山大学 | Method and system for sequencing network MP3 (moving picture experts group audio layer-3) tone qualities |
| CN106230619B (en) * | 2016-07-21 | 2019-12-17 | 湖南智卓创新信息产业股份有限公司 | Data sending and receiving method and device, and data transmission method and system |
| CN108512708B (en) * | 2017-02-24 | 2023-01-06 | 中兴通讯股份有限公司 | Cache calculation method and device |
| CN110213170A (en) * | 2018-02-28 | 2019-09-06 | 华为技术有限公司 | A kind of method and device of message transmissions |
| CN112217842A (en) * | 2019-07-09 | 2021-01-12 | 北京声智科技有限公司 | Data transmission method and device |
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2005
- 2005-04-28 CN CNB200510034465XA patent/CN100542132C/en not_active Expired - Fee Related
Non-Patent Citations (5)
| Title |
|---|
| Internet上的语音传输技术探讨. 蒋先刚.计算机工程与应用，第38卷第24期. 2002 |
| Internet上的语音传输技术探讨. 蒋先刚.计算机工程与应用，第38卷第24期. 2002 * |
| 一种多TCP连接的建模分析. 林宇 |
| 一种多TCP连接的建模分析. 林宇；程时端.电子与信息学报，第24卷第12期. 2002 * |
| 程时端.电子与信息学报，第24卷第12期. 2002 |
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| CN1855878A (en) | 2006-11-01 |
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