CN102394993A - Method for automatically adjusting and improving RTP (Realtime Transport Protocol) flow quality based on voice coding in VoIP (Voice over Internet Portocol) network - Google Patents
Method for automatically adjusting and improving RTP (Realtime Transport Protocol) flow quality based on voice coding in VoIP (Voice over Internet Portocol) network Download PDFInfo
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- CN102394993A CN102394993A CN2011103414274A CN201110341427A CN102394993A CN 102394993 A CN102394993 A CN 102394993A CN 2011103414274 A CN2011103414274 A CN 2011103414274A CN 201110341427 A CN201110341427 A CN 201110341427A CN 102394993 A CN102394993 A CN 102394993A
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Abstract
The invention relates to a method for automatically adjusting and improving RTP (Realtime Transport Protocol) flow quality based on voice coding in a VoIP (Voice over Internet Portocol) network. In the method, fields related to RTP flow transmission parameters are extracted from a statistic report of a RTCP (Real Time Control Protocol) conversation in RTP flow transmission process, and conversion conditions among different compression coding methods are set; when the related fields satisfy the conversion conditions, a new RTP flow compression coding method is converted by a signaling consultation so as to realize the dynamic adjustment of the coding to reduce network delay and jitter under an unstable network state by RTCP statistic information, thereby improving the quality of voice and videos, and further improving the satisfaction of customers using a VoIP; and in the method, the application way is simple and convenient, the cost is low, and the application range is wider.
Description
Technical field
The present invention relates to network communications technology field, particularly speech coding technology field in the voip network specifically is meant in a kind of voip network and adjusts raising rtp streaming method for quality automatically based on speech coding.
Background technology
VoIP (Voice over Internet Protocol) be with the simulation sound signal after overcompression and package; Under the environment of IP network, carry out the transmission of speech sound signal with the form of data packet, be commonly called Internet Protocol telephone, the networking telephone or be called for short IP phone.
The basic principle of VoIP is: the compression algorithm through voice is carried out processed compressed to encoded speech data; Pack these speech datas by the TCP/IP standard then; Deliver to reception ground to packet through IP network, string together these VoPs again, through after the decompression processing; Revert to original voice signal, thereby reach the purpose that transmits voice by the Internet.The core of IP phone and key equipment are IP gateways, and it is mapped as corresponding regional gateway ip address to the each department area code.These information leave in the database, and the data process software that continues will be accomplished functions such as call treatment, digital speech packing, routing management.
When the user dialed toll telephone, gateway was confirmed the IP address of respective gateway according to the area code database data; And with in this IP address adding IP packet; Select best route simultaneously, to reduce propagation delay time, the IP packet arrives the gateway of destination through Internet.Area in that some Internet do not extend to as yet or temporarily do not set up gateway can be provided with route, through the long-distance network switching, realizes communication service by nearest gateway.
At the multimedia service based on Internet, in teleconference, monitoring remote video, data often will be through overcompression in transmission over networks.And the video/audio business on the network realizes through RTP/RTCP mostly.RTP host-host protocol based on UDP does not have the such security mechanism of TCP; And the real-time of video data requires to occur the re-transmission that data error code or packet loss do not allow data yet; Therefore, the quality of multimedia transmission is the lower main cause of voip network user satisfaction, wherein network delay always; Network jitter, Network Packet Loss are the modal problems that influences voice quality.So, need provide a kind of and dynamically adjust coding to network conditions, improve the method for multimedia transmission quality.
Summary of the invention
The objective of the invention is to have overcome above-mentioned shortcoming of the prior art; Provide a kind of can be according to the Network Transmission situation; Real-time and Dynamic adjustment speech coding to be adapting to the Network Transmission requirement, and then improves multimedia transmission quality, and application mode is easy; Realize with low costly, and range of application is adjusted automatically based on speech coding in the voip network comparatively widely and is improved the rtp streaming method for quality.
In order to realize above-mentioned purpose, adjust raising rtp streaming method for quality automatically based on speech coding in the voip network of the present invention and may further comprise the steps:
(1) in voip network, connects after being connected between calling terminal and terminal called, consult to confirm initial realtime transmission protocol RTP stream compression coding mode and RTP Control Protocol RTCP session cycle through the signaling between calling terminal and terminal called;
(2) between described calling terminal and terminal called, carry out rtp streaming transmission and RTCP flow transmission according to determined rtp streaming compression coding mode and RTCP session cycle;
(3) in the media flow transmission process, from the statistical report of described RTCP session, extract the field relevant with the rtp streaming transmission parameter;
(4), the switch condition between the different compression coded system is set according to the field of being extracted;
(5) when relevant field satisfies described switch condition, consult to adopt new rtp streaming compression coding mode to carry out the rtp streaming transmission through signaling between described calling terminal and terminal called, and return step (2).
Automatically adjust in the raising rtp streaming method for quality based on speech coding in this voip network, the switch condition between described different compression coded system is specially: the corresponding relation between different compression coded system and the rtp streaming transmission parameter.
Automatically adjust in the raising rtp streaming method for quality based on speech coding in this voip network, the described field relevant with the rtp streaming transmission parameter comprises: arrival interval shake, packet loss, time-delay and Loop Round Trip Time.
Automatically adjust in the raising rtp streaming method for quality based on speech coding in this voip network, the corresponding relation between described different compression coded system and the rtp streaming transmission parameter is specially: the corresponding relation between different compression coded system and arrival interval dithering threshold, packet loss threshold value, delay threshold and the Loop Round Trip Time threshold value.
Automatically adjust in the raising rtp streaming method for quality based on speech coding in this voip network, the described RTCP session cycle is 5 seconds.
Adopted in the voip network of this invention to adjust automatically and improved the rtp streaming method for quality based on speech coding; In the rtp streaming transmission course; From the statistical report of described RTCP session, extract the field relevant, and the switch condition between the different compression coded system is set with the rtp streaming transmission parameter; When relevant field satisfies described switch condition; Consult the new rtp streaming compression coding mode of conversion through signaling and carry out the rtp streaming transmission, thereby realize utilizing the RTCP statistical information, under the state of unstable networks; Dynamically adjustment is encoded; Reduce the problem of network delay and network jitter, improve the voice and video quality, and then improve the CSAT that VoIP uses.And adjust automatically based on speech coding in the voip network of the present invention that to improve rtp streaming method for quality application mode easy, realize with low costly, and range of application is comparatively extensive.
Description of drawings
Fig. 1 is for adjust the flow chart of steps that improves the rtp streaming method for quality based on speech coding in the voip network of the present invention automatically.
Embodiment
In order more to be expressly understood technology contents of the present invention, the special following examples of lifting specify.
See also shown in Figure 1, for adjusting the flow chart of steps that improves the rtp streaming method for quality automatically based on speech coding in the voip network of the present invention.
In one embodiment, adjusting raising rtp streaming method for quality automatically based on speech coding in this voip network may further comprise the steps:
(1) in voip network, connects after being connected between calling terminal and terminal called, consult to confirm initial realtime transmission protocol RTP stream compression coding mode and RTP Control Protocol RTCP session cycle through the signaling between calling terminal and terminal called;
(2) between described calling terminal and terminal called, carry out rtp streaming transmission and RTCP flow transmission according to determined rtp streaming compression coding mode and RTCP session cycle;
(3) in the media flow transmission process, from the statistical report of described RTCP session, extract the field relevant with the rtp streaming transmission parameter;
(4), the switch condition between the different compression coded system is set according to the field of being extracted;
(5) when relevant field satisfies described switch condition, consult to adopt new rtp streaming compression coding mode to carry out the rtp streaming transmission through signaling between described calling terminal and terminal called, and return step (2).
In a kind of more preferably execution mode, the switch condition between described different compression coded system is specially: the corresponding relation between different compression coded system and the rtp streaming transmission parameter.The described field relevant with the rtp streaming transmission parameter comprises: arrival interval shake, packet loss, time-delay and Loop Round Trip Time.Corresponding relation between then described different compression coded system and the rtp streaming transmission parameter is specially: the corresponding relation between different compression coded system and arrival interval dithering threshold, packet loss threshold value, delay threshold and the Loop Round Trip Time threshold value.
In a kind of preferred execution mode, the described RTCP session cycle is 5 seconds.
In practical application, in the process of utilizing voip network to converse, accomplish the negotiation of rtp streaming coding through using the signaling negotiation method.The process of consulting is based on the network capabilities of both call sides, and the encoding list that provides preferentially mates, and coupling goes up coding and is used as the result of negotiation at first, and finally uses this rtp streaming to encode to transmit rtp streaming.The of the present invention adjustment automatically based on speech coding improved the rtp streaming method for quality; It is through the RTCP statistical information; Be implemented under the state of unstable networks, dynamically adjustment is encoded, to reduce the problem of network delay and network jitter; Improve the voice and video quality of conversation, guarantee the CSAT that VoIP uses.
Caller and called after connection is with the rtp streaming of one type of a kind of coding transmission.Existing is that of the present invention the adjustment automatically based on speech coding of example explanation improved the step of rtp streaming method for quality in practical application with the audio coding, specific as follows:
At first communicating pair is opened the transmission of the medium control flows of RTCP when utilizing the RTP media stream.
1, before transmission course, suitable adjustment (according to various network conditions) is done in the adjustment of RTCP session cycle, be defaulted as 5s, consult to carry out media flow transmission through signaling, the result who supposes negotiation is the G711 coded system;
2, in transmission course; Unpack field from RTCP (RR=201) statistical report of receiving; Interarrival jitter (arrival interval shake), the threshold value of loss fraction (packet loss) and the corresponding relation between the respective coding mode are set, and described corresponding relation is as shown in the table:
| Coded system | The interarrival threshold value | Loss fraction threshold value |
| G711 | 50 | ?1 |
| G729 | 100 | ?3 |
| G723 | 150 | ?7 |
| … | … | ?… |
The shake of table 1 arrival interval and the threshold value of packet loss and the mapping table of respective coding mode
Same, in this step, can actual packet loss be foundation to the valve limit value also with time-delay, coded system is adjusted, till voice quality is satisfied;
When 3, the above-mentioned field in receiving the RTCP bag is crossed threshold threshold; Initiate reinvite and invite signaling or update renewal signaling (is example with sip) to consult the media session coded system again, as after reaching the valve limit of G723 coded system, preferred G723 coding is held consultation; Reach the purpose that reduces the network bandwidth; Reduce network delay, network jitter and packet loss, the purpose of raising voice quality;
4, when using higher compressed encoding, receive that the threshold value in the RTCP bag is less, when being lower than the threshold value of G711 coded system; Again initiate coding negotiation; Optimized encoding is held consultation for the G711 coded system, thereby reduces the voice distortion that coding brings, and improves voice quality;
5, same, (Loop Round Trip Time, the threshold values of RR time of receipt (T of R)-LSR-DLSR) arrives above-mentioned effect to the coding dynamic negotiation also can to use the RTT value in this method.
Adopted in the voip network of this invention to adjust automatically and improved the rtp streaming method for quality based on speech coding; In the rtp streaming transmission course; From the statistical report of described RTCP session, extract the field relevant, and the switch condition between the different compression coded system is set with the rtp streaming transmission parameter; When relevant field satisfies described switch condition; Consult the new rtp streaming compression coding mode of conversion through signaling and carry out the rtp streaming transmission, thereby realize utilizing the RTCP statistical information, under the state of unstable networks; Dynamically adjustment is encoded; Reduce the problem of network delay and network jitter, improve the voice and video quality, and then improve the CSAT that VoIP uses.And adjust automatically based on speech coding in the voip network of the present invention that to improve rtp streaming method for quality application mode easy, realize with low costly, and range of application is comparatively extensive.
In this specification, the present invention is described with reference to its certain embodiments.But, still can make various modifications and conversion obviously and not deviate from the spirit and scope of the present invention.Therefore, specification and accompanying drawing are regarded in an illustrative, rather than a restrictive.
Claims (5)
1. adjust raising rtp streaming method for quality automatically based on speech coding in a voip network, it is characterized in that described method may further comprise the steps:
(1) in voip network, connects after being connected between calling terminal and terminal called, consult to confirm initial realtime transmission protocol RTP stream compression coding mode and RTP Control Protocol RTCP session cycle through the signaling between calling terminal and terminal called;
(2) between described calling terminal and terminal called, carry out rtp streaming transmission and RTCP flow transmission according to determined rtp streaming compression coding mode and RTCP session cycle;
(3) in the media flow transmission process, from the statistical report of described RTCP session, extract the field relevant with the rtp streaming transmission parameter;
(4), the switch condition between the different compression coded system is set according to the field of being extracted;
(5) when relevant field satisfies described switch condition, consult to adopt new rtp streaming compression coding mode to carry out the rtp streaming transmission through signaling between described calling terminal and terminal called, and return step (2).
2. adjust raising rtp streaming method for quality automatically based on speech coding in the voip network according to claim 1, it is characterized in that the switch condition between described different compression coded system is specially:
Corresponding relation between different compression coded system and the rtp streaming transmission parameter.
3. adjust raising rtp streaming method for quality automatically based on speech coding in the voip network according to claim 2, it is characterized in that the described field relevant with the rtp streaming transmission parameter comprises: arrival interval shake, packet loss, time-delay and Loop Round Trip Time.
4. adjust raising rtp streaming method for quality automatically based on speech coding in the voip network according to claim 3, it is characterized in that the corresponding relation between described different compression coded system and the rtp streaming transmission parameter is specially:
Corresponding relation between different compression coded system and arrival interval dithering threshold, packet loss threshold value, delay threshold and the Loop Round Trip Time threshold value.
5. according to adjusting raising rtp streaming method for quality automatically based on speech coding in each described voip network in the claim 1 to 4, it is characterized in that the described RTCP session cycle is 5 seconds.
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| CN103259945A (en) * | 2013-05-20 | 2013-08-21 | 苏州洲际传书信息科技有限公司 | Method for achieving speech coding dynamic switching in voip conversation according to network condition |
| CN103856461A (en) * | 2012-12-04 | 2014-06-11 | 联芯科技有限公司 | Consultative adjustment method of IMS service real-time media streams |
| CN105764093A (en) * | 2014-12-19 | 2016-07-13 | 中国移动通信集团黑龙江有限公司 | Method and device for selecting data compression level |
| CN108924113A (en) * | 2018-06-25 | 2018-11-30 | 京信通信系统（中国）有限公司 | Method of adjustment, device, computer storage medium and the equipment of speech encoding rate |
| CN109729318A (en) * | 2019-01-07 | 2019-05-07 | 深圳英飞拓科技股份有限公司 | Video data playback process method, apparatus, computer equipment and storage medium |
| CN111164947A (en) * | 2017-08-14 | 2020-05-15 | 英国电讯有限公司 | Method and device for encoding audio and/or video data |
| CN113452723A (en) * | 2021-08-31 | 2021-09-28 | 深圳鼎信通达股份有限公司 | Voice processing method, device and storage medium |
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