CN101056283B - Voice gateway and method for providing VoIP service - Google Patents
Voice gateway and method for providing VoIP service Download PDFInfo
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- CN101056283B CN101056283B CN200710110622XA CN200710110622A CN101056283B CN 101056283 B CN101056283 B CN 101056283B CN 200710110622X A CN200710110622X A CN 200710110622XA CN 200710110622 A CN200710110622 A CN 200710110622A CN 101056283 B CN101056283 B CN 101056283B
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Abstract
The invention belongs to the network communication area, discloses the voice gateway which can provide VoIP service, and method for providing the VoIP service based on the voice gateway. The voice gateway includes the PSTN entity, VoIP entity, selection unit and IP availability analysis unit, wherein the IP availability analysis unit is used to detect the network quality of the IP network via sending the detection packet periodically; or the voice gateway includes the PSTN entity, VoIP entity, selection unit and RTP quality monitoring unit, wherein the RTP quality monitoring unit is used to real time monitoring the service quality of the RTP (read time transport protocol) via the RTCP (real time transport control protocol). So the selection unit can determine the type of the call to originate according the network quality; and/or the selection unit can determine whether to switch the call according to the network quality or the RTP service quality in the VoIP call.
Description
Technical field
The present invention relates to communication technical field, relate in particular to the voice gateways that the VoIP business can be provided, and the method that the VoIP business is provided that is applied thereon.
Background technology
VoIP (Voice Over IP) is with IP (Internet Protocol, Internet protocol) packet switching network is a transmission platform, to analog voice signal compress, a series of special processings such as packing, make it the technology that can adopt connectionless IP packet data package to transmit.Utilizing voip technology to carry out IP phone on IP-based data communication networks such as intranet, wide area network, Internet uses.The cost of the phone call of dialing the world, national distance by IP phone is cheap, can reduce the conversation cost greatly, guarantees good voice quality simultaneously.Yet the voice quality of VoIP is subjected to the influence of IP network quality bigger, and the main factor that wherein influences voice quality is time-delay, shake and the packet loss of IP network.
Existing voip technology is generally called out the selection of initiation by the priority that goes up the configuration entity in voice gateways (Voice Gateway).Such as, two identical numbers of entity coupling of configuration on voice gateways, one is used PSTN (Public Switched Telephone Network, public switch telephone network) to call out, and one is used voip call.If the priority height that the PSTN entity is provided with is then called out and is used the PSTN network; If the priority height that the VoIP entity is provided with is then called out and is used IP network.Also have a kind of situation, if IP network is unavailable, even it is provided with high priority, voice gateways are still selected the PSTN entity for use when making a call.
Whether there are following two major defects in above-mentioned prior art: 1, can not make a call according to the decision of IP network quality.For example, suppose that VoIP is provided with high priority,, call out and still can use IP network even then IP network quality (as time-delay and shake) can not satisfy the call condition of VoIP.2, can not dynamically switch, just in the VoIP communication process, can't switch to the original calling that continues of PSTN network from disabled IP network.Particularly, if IP network has become unavailablely in the VoIP communication process, existing voice gateways technology then will stop the conversation of this road, make a call again thereby make the user manually to dial PSTN.
Summary of the invention
In view of this, the object of the invention is to provide a kind of voice gateways that can dynamically perception IP network quality, thereby can be to initiate voip call or initiate PSTN and call out and can network quality in time initiate the voip call that PSTN calls out the connecting quality difference in the voip call process according to the network quality decision in the time need making a call.
Another purpose of the present invention is to provide the method that the VoIP business is provided accordingly, to support the application of above-mentioned voice gateways.
In order to achieve the above object, the invention discloses a kind of voice gateways.Described voice gateways comprise PSTN entity, VoIP entity, selected cell and IP available analyses unit.Wherein: described PSTN entity is used for carrying out PSTN by the PSTN network and calls out; Described VoIP entity is used for carrying out voip call by IP network; Described IP available analyses unit is used for detecting by regular transmission probe messages the network quality of IP network, and testing result is sent to described selected cell; Described selected cell is used for selecting the used entity that makes a call according to the network quality that described IP available analyses unit obtains.Particularly, when needs make a call, described selected cell at first detects in real time the described network quality that obtains according to described IP available analyses unit and judges whether described IP network satisfies the requirement of initiating VoIP, if judged result is for being then determine to use described VoIP entity to make a call.
For above-mentioned voice gateways, preferably, it also comprises RTP (Real-time TransportProtocol, RTP) quality monitor unit.Described RTP quality monitor unit is used for the RTP service quality that recipient's report by RTCP (Real-Time Transport Control Protocol, RTCP Real-time Transport Control Protocol) and Sender Report are monitored described IP network.Like this, in the voip call process, described selected cell just can determine whether using described PSTN entity to initiate PSTN according to described RTP service quality and call out, with the former service quality voip call of variation that continues.
For above-mentioned voice gateways, preferably, it also includes the network quality statistical form.Described network quality statistical form is used to write down the network quality that described IP available analyses unit draws.In the time will making a call, described selected cell is at first checked described network quality statistical form, if wherein record shows that described IP network satisfies the requirement of initiating voip call, then described selected cell decision uses described VoIP entity to make a call.
For above-mentioned voice gateways, preferably, it also includes timer.Described timer is used to calculate from sending described probe messages to the stand-by period of receiving its response message.In the voip call process, if described timer shows that exceed the scheduled wait time does not receive that yet the response message of described probe messages, then described selected cell decision use the described PSTN former voip call that continues.
For above-mentioned voice gateways, preferably, it also is provided with the first threshold and second threshold value.In the voip call process: when described RTP service quality was lower than described first threshold, described selected cell decision used described PSTN entity to be communicated with described voice gateways and the other side's voice gateways; When described RTP service quality was lower than described second threshold value, described selected cell decision was enabled the PSTN circuit that has been communicated with and is called out former voip call is switched to PSTN.
Simultaneously, in order to achieve the above object, the invention also discloses another kind of voice gateways.Described voice gateways comprise PSTN entity, VoIP entity, selected cell and RTP quality monitor unit.Wherein: described PSTN entity is used for carrying out PSTN by the PSTN network and calls out; Described VoIP entity is used for carrying out voip call by IP network; Described RTP quality monitor unit is used for the RTP service quality that recipient's report by RTCP and Sender Report are monitored described IP network, and monitoring result is sent to described selected cell.Like this, in the voip call process, described selected cell just can determine whether using described PSTN entity to initiate PSTN according to described RTP service quality and call out, with the service quality voip call of variation that continues.
For above-mentioned voice gateways, preferably, it also is provided with the first threshold and second threshold value.In the voip call process: when described RTP service quality was lower than described first threshold, described selected cell decision used described PSTN entity to be communicated with described voice gateways and the other side's voice gateways; When described RTP service quality was lower than described second threshold value, described selected cell decision was enabled the PSTN circuit that has been communicated with and is called out former voip call is switched to PSTN.
In order to reach above-mentioned another purpose, a kind of method of the VoIP of providing business is provided, it includes the following step:
Steps A 1, local terminal gateway regularly send probe messages to the opposite end gateway;
Steps A 4, when needs made a call, described local terminal gateway judged that whether described network quality satisfies the predetermined requirement of initiating voip call, if judged result is for being then initiate voip call.
For the above-mentioned method that the VoIP business is provided, preferably, in the voip call process, it also includes the following step:
Step B1, the RTP service quality that described local terminal gateway is monitored described IP network by recipient's report and the Sender Report of RTCP;
Step B2, if described local terminal gateway detects described RTP service quality when being lower than presetting first threshold, then the backstage is communicated with the PSTN circuit;
Step B3 if described local terminal gateway detects described RTP service quality when being lower than the second default threshold value, then enables the PSTN circuit that has been communicated with the former calling that continues, and stops former voip call simultaneously.
For the above-mentioned method that the VoIP business is provided, preferably, in the voip call process, it also includes the following step:
Steps A 5, described local terminal gateway judge from sending described probe messages to the stand-by period of receiving its response message whether exceed the scheduled wait time, if judged result is for being then initiate the PSTN calling with the former voip call that continues.
For the above-mentioned method that the VoIP business is provided, preferably, described probe messages is the UDP message, and described response message has been stamped timestamp.
Simultaneously, in order to reach above-mentioned another purpose, the invention also discloses the method that another kind provides the VoIP business.Described method by the RTP service quality that recipient's report and the Sender Report of RTCP are monitored IP network, makes the local terminal gateway whether to initiate PSTN according to described RTP service quality decision and calls out the former voip call that continues in the voip call process.
For the above-mentioned method that the VoIP business is provided, preferably, mainly include the following step:
Step B1, the RTP service quality that described local terminal gateway is monitored described IP network by recipient's report and the Sender Report of RTCP;
Step B2, if described local terminal gateway detects described RTP service quality when being lower than presetting first threshold, then the backstage is communicated with the PSTN circuit;
Step B3 if described local terminal gateway detects described RTP service quality when being lower than the second default threshold value, then enables the PSTN circuit that has been communicated with the former calling that continues, and stops former voip call simultaneously.
Technical solution of the present invention has following beneficial effect: 1, in the time will making a call, it still is the PSTN network that IP network is used in the calling that can be according to the network quality of the IP network corresponding decision of the present invention is initiated; 2, in the voip call process, if IP network is unavailable, the present invention can automatically switch to calling on the PSTN circuit, thereby has guaranteed that original calling is not interrupted; 3, the present invention does not relate to the architecture of IP phone, only relates to the voice gateways resolution problem, thereby can support H.323/SIP (Session Initiation Protocol, session initiation protocol) agreement simultaneously.
Description of drawings
Fig. 1 is the structured flowchart of voice gateways preferred embodiment of the present invention;
Fig. 2 the invention provides the system schematic of one of method embodiment of VoIP business for application;
Fig. 3 is the flow chart that the invention provides one of the method embodiment of VoIP business;
Fig. 4 is for using two the system schematic of the method embodiment the invention provides the VoIP business;
Fig. 5 is two the flow chart that the invention provides embodiment in the method for VoIP business.
Embodiment
The present invention mainly conceives and is by dynamic perception IP network quality, makes it possible to select suitable substance P STN entity or VoIP entity to call out to initiate and/or go up in time with it when the IP network degradation voip call according to the IP network quality and switches to the PSTN calling and need not interruption.Below in conjunction with accompanying drawing in detail it is described in detail.
Fig. 1 has provided the structured flowchart of voice gateways preferred embodiment provided by the present invention.As shown in Figure 1, voice gateways 100 provided by the present invention comprise PSTN entity 110, VoIP entity 120, selected cell 130, IP available analyses unit 140 and RTP quality monitor unit 150.And the signal transitive relation between each assembly includes among Fig. 1: represent that 1. PSTN calls out; 2. represent voip call; 3. represent probe messages, 4. expression and 3. corresponding response message; 5. represent the RTP message, be mainly used in to voip call provides the end to end network transmission and serve; 6. represent the RTCP message, comprise " recipient " report and " sender " report, be used for the service quality and the network congestion degree of monitoring RTP; 7. the network quality of representing IP network 300 relates generally to time-delay, shake, and the performance parameter value such as packet loss of IP network 300; 8. the RTP service quality of representing IP network 300 relates generally to the RTP service quality and the network congestion degree of IP network 300 in the voip call process; 9. presentation-entity is selected information.
Wherein, PSTN entity 110 and VoIP entity 120 can realize that key of the present invention is the proposition of IP available analyses unit 140 and RTP quality monitor unit 150 and the adaptations of selected cell 130 with arbitrary known execution mode in the prior art.
Need to prove, the most preferred embodiment as voice gateways of the present invention shown in Figure 1, it includes IP available analyses unit 140 and RTP quality monitor unit 150 simultaneously.But this does not also mean that IP available analyses unit 140 and the 150 inevitable appearance simultaneously of RTP quality monitor unit.In actual applications, those skilled in the art can understand on the basis of the principle of the invention fully, the suboptimum embodiment that implements out only to include IP available analyses unit 140 or only include RTP quality monitor unit 150.
Thus, below IP available analyses unit 140 and RTP quality monitor unit 150 are considered as separate assembly, both technology realize that details is introduced to this respectively.
At first, introduce the working mechanism of IP available analyses unit 140 and the content of operation of selected cell 130 corresponding IP available analyses unit 140 in detail with reference to Fig. 1,2 and 3.
As shown in Figure 2, the IP available analyses unit 140 among the local terminal gateway 100#1 at a certain time interval (configurable) send probe messages 3. to opposite end gateway 100#2.In general, 3. probe messages is preferably UDP (User Datagram Protocol, User Datagram Protoco (UDP)) message, and do not recommend to use ICMP (Internet Control Messages Protocol, the Internet Internet Control Message Protocol) message, this is because many fire compartment walls limit the ICMP agreement.Opposite end gateway 100#2 receives and can respond after probe messages 3., and 4. response message is sent back to local terminal gateway 100#1.Like this, IP available analyses unit 140 among the local terminal gateway 100#1 after receiving that response message 4., just can according to probe messages 3. and response message 4. calculate between local terminal gateway 100#1 and the opposite end gateway 100#2 IP network 300 for the time of the time-delay and the shake of UDP message.
Simultaneously, each detection can send a plurality of probe messages by configuration and 3. test, thereby makes IP available analyses unit 140 among the local terminal gateway 100#1 can just can calculate the packet loss of IP network 300 according to the response message that returns number 4..Certainly, the probe messages that each detection sends is 3. many more, and the network quality that IP available analyses unit 140 records is 7. accurate more.
Based on the consideration of conveniently consulting, 7. the network quality that IP available analyses unit 140 can be recorded is stored in the table among the local terminal gateway 100#1 in chronological order, is called the network quality statistical form for the time being.Like this, when local terminal gateway 100#1 need make a call, selected cell 130 will at first be checked the state-of-the-art record of storing in the network quality statistical form on it.If this state-of-the-art record shows IP network 300 and satisfies the requirement of initiating voip call that then selected cell 130 will determine to use VoIP entity 120 to initiate voip call 2., call out 1. otherwise selected cell 130 will determine to use PSTN entity 110 to initiate PSTN.
In addition, 2. in time switch on the PSTN network 200 in order IP network 300 disabled voip calls in time to have occurred, local terminal gateway 100#1 also can utilize timer to calculate to send probe messages from IP available analyses unit 140 and 3. receive the respective response message stand-by period 4. to it.Like this, in voip call 2. in the process, if timer shows that exceeding the scheduled wait time does not 4. receive probe messages response message 3. yet, then selected cell 130 will use PSTN entity 110 to initiate the PSTN calling 1. at once, 2. need not interrupt call with the former voip call that continues, also promptly need not to require the user to dial again and initiate the PSTN calling 1..
And,,, therefore can provide comfort noise, thereby make the user avoid by IP network in the unavailable calling handoff procedure that causes at this because of complete quiet hanging up the telephone by the local terminal gateway because the voip call data that can't transmit the IP network 300 of need flowing through are 2..
By above-mentioned introduction as can be known, provide method corresponding to the VoIP business that can be applicable on the voice gateways 100 provided by the present invention, the flow chart of wherein 7. calling out initiation according to the network quality of IP network 300 and switching mainly includes the following step as shown in Figure 3:
Steps A 1, local terminal gateway 100#1 regularly sends probe messages 5. to opposite end gateway 100#2;
Steps A 2, opposite end gateway 100#2 receives after probe messages is 5. and responds, and 6. response message is sent back to local terminal gateway 100#1;
Steps A 3, local terminal gateway 100#1 receives after the response message, to probe messages 5. and response message 6. carry out statistical analysis, 7. the network quality that draws IP network 300 between local terminal gateway 100#1 and the opposite end gateway 100#2 mainly includes time-delay, shake, and the packet loss etc. of IP network 300;
Steps A 4, when local terminal gateway 100#1 need make a call, 7. whether its network quality that will at first judge IP network satisfy the predetermined requirement of initiating voip call, if judged result is for being then initiate voip call, if judged result be otherwise initiation PSTN calling;
Steps A 5, in the voip call process, local terminal gateway 100#1 will continue to judge from sending probe messages 5. to receiving whether its response message stand-by period 6. exceeds the scheduled wait time, if judged result is for being then initiate the former voip call of PSTN call proceeding and need not to interrupt, if judged result for otherwise do not carry out any operation.
In a word, the network quality that regularly sends IP network 300 between probe messages real-time perception local terminal voice gateways 100#1 and the opposite end voice gateways 100#2 by the whole duration of work utilization at local terminal voice gateways 100#1 7., the invention enables local terminal voice gateways 100#1 can be when needs make a call be to initiate voip call or initiate PSTN and call out according to the 7. corresponding decision of network quality of IP network 300, and in the voip call process, when 7. the network quality of IP network becomes unavailable, in time initiate the former calling of PSTN call proceeding and need not interruption.
Then, introduce the working mechanism of RTP quality monitor unit 150 and the content of operation of selected cell 130 corresponding RTP quality monitor unit 150 in detail with reference to Fig. 1,4 and 5.
As shown in Figure 4, in voip call 2. in the process, the RTP quality monitoring unit 150 among the local terminal gateway 100#1 by " recipient's report " and " Sender Report " realization in 5. of RTCP message to the service quality of RTP and the real-time detection of network congestion degree.
RTP (Real-time Transport Protocol, RTP) issues as RFC1889 by IETF (Internet engineering duty group), its be defined in one to one or the transmission situation of one-to-many under work, its objective is provides temporal information and realizes that stream is synchronously.RTP itself only guarantees the transmission of real time data, can not provide reliable transfer mechanism for transfer data packets in order, flow control or congested control are not provided yet, and it relies on RTCP (Real-Time Transport ControlProtocol, RTCP Real-time Transport Control Protocol) that these services are provided.That is to say that during the RTP session, each participant periodically transmits the RTCP message, comprise the quantity of data packets that has sent, the statistics of losing such as quantity of data packets in the RTCP message.
Like this, participate in RTP quality monitor unit 150 in the voice gateways (comprising local terminal gateway 100#1 and opposite end gateway 100#2) of voip call by rationally utilizing the statistical information of RTCP message in 5., can detect the RTP service quality of IP network 300 and network congestion degree etc. in real time.Thereby make, 2. carry out in the process in voip call, 8. selected cell 130 can judge whether IP network 300 the network quality variation has taken place according to the RTP service quality that RTP quality monitor unit 150 records, if judged result is for being then determine to select for use PSTN entity 110 to initiate PSTN to call out 1., call out 1. 2. voip call is switched to PSTN, and need not to interrupt original calling.
Certainly, in order effectively to reduce the problem of switching required time and effectively preventing to cause to switch repeatedly of calling out because of network quality is unstable, usually be provided with two threshold values in the voice gateways 100 of the present invention, wherein first threshold is used for judging that whether trigger the backstage enables PSTN side resource, second threshold value and be used for judging whether to trigger and switch to PSTN.
Thereby make, when 8. RTP service quality be lower than first threshold, 130 of selected cells will use PSTN entity 110 to be communicated with PSTN network 200 between local terminal gateway 100#1 and the opposite end gateway 100#2, make opposite end 100#2 be equivalent to have one road PSTN to call out to arrive and be held; And 8. continue to reduce so that when being lower than second threshold value when RTP service quality, 130 of selected cells will be enabled the PSTN network 200 that has been communicated with and carry out calling line and switch, and also promptly stop original voip call and it will be routed on the PSTN network 2. the time.
Similarly, corresponding to the method that the VoIP business is provided that can be applicable on the voice gateways 100 provided by the present invention, wherein the flow chart of 8. calling out switching according to the RTP service quality of IP network 300 mainly includes the following step as shown in Figure 5:
Step B1, in the voip call process, local terminal gateway 100#1 by RTCP recipient's report and Sender Report monitor IP network 300 between local terminal gateway 100#1 and the opposite end gateway 100#2 in real time RTP service quality 8.;
Step B2, when 8. the RTP service quality that detects IP network 300 as local terminal gateway 100#1 be lower than presetting first threshold, then the backstage is communicated with a PSTN circuit by the PSTN network between local terminal gateway 100#1 and the opposite end gateway 10,0#2 200, also promptly makes local terminal gateway 100#1 have one road PSTN to call out to arrive opposite end gateway 100#2 and is maintained by opposite end gateway 100#2;
Step B3,8. the RTP service quality that detects IP network 300 as local terminal gateway 100#1 continue to be reduced to when being lower than the second default threshold value, then enable the above-mentioned PSTN circuit that has been communicated with, also promptly initiate the former voip call of PSTN call proceeding, stop former voip call simultaneously.
In a word, by in the voip call process, utilizing RTP service quality that RTCP monitors IP network 300 between local terminal voice gateways 100#1 and the opposite end gateway 100#2 in real time 8., the invention enables local terminal voice gateways 100#1 8. not satisfy and in time call out switching when VoIP requires in the RTP of IP network 300 service quality.And in order to reduce the switching repeatedly of switching the required time and preventing to cause because of network jitter, the present invention also provides the preferred embodiment of default two threshold values.
What need statement is that foregoing invention content and embodiment are intended to prove the practical application of technical scheme provided by the present invention, should not be construed as the qualification to protection range of the present invention.Those skilled in the art are in spirit of the present invention and principle, when doing various modifications, being equal to and replacing or improve.Protection scope of the present invention is as the criterion with appended claims.
Claims (9)
1. voice gateways comprise PSTN entity, VoIP entity and selected cell; It is characterized in that, also comprise IP available analyses unit, be used for detecting by regular transmission probe messages the network quality of IP network, and testing result is sent to described selected cell, described selected cell is called out the selection of initiating used entity according to described network quality; The RTP quality monitor unit, be used for the RTP service quality that recipient's report by RTCP and Sender Report are monitored described IP network, make described selected cell can determine whether to use the former voip call that continues of described PSTN entity according to described RTP service quality;
Also be provided with the first threshold and second threshold value, in the voip call process:
When described RTP service quality was lower than described first threshold, described selected cell decision used described PSTN entity to be communicated with described voice gateways and the other side's voice gateways;
When described RTP service quality was lower than described second threshold value, described selected cell decision was enabled the PSTN circuit that has been communicated with and is called out former voip call is switched to PSTN.
2. voice gateways as claimed in claim 1 is characterized in that, also include the network quality statistical form, are used to write down the network quality that described IP available analyses unit draws;
In the time will making a call, described selected cell is at first checked described network quality statistical form, if wherein record shows that described IP network satisfies the requirement of initiating voip call, then described selected cell decision uses described VoIP entity to make a call.
3. voice gateways as claimed in claim 1 is characterized in that, also include timer, are used to calculate from sending described probe messages to the stand-by period of receiving its response message;
In the voip call process, if described timer shows that exceed the scheduled wait time does not receive that yet the response message of described probe messages, then described selected cell decision use the former voip call that continues of described PSTN entity.
4. voice gateways comprise PSTN entity, VoIP entity and selected cell; It is characterized in that, also comprise the RTP quality monitor unit, be used for the RTP service quality that recipient's report by RTCP and Sender Report are monitored IP network, and monitoring result sent to described selected cell, whether described selected cell uses the former voip call that continues of described PSTN entity according to described RTP service quality decision;
Also be provided with the first threshold and second threshold value, in the voip call process:
When described RTP service quality was lower than described first threshold, described selected cell decision used described PSTN entity to be communicated with described voice gateways and the other side's voice gateways;
When described RTP service quality was lower than described second threshold value, described selected cell decision was enabled the PSTN circuit that has been communicated with and is called out former voip call is switched to PSTN.
5. the method that the VoIP business is provided is characterized in that, includes the following step:
Steps A 1, local terminal gateway regularly send probe messages to the opposite end gateway;
Steps A 2 responds after described opposite end gateway is received described probe messages, and response message is sent back to described local terminal gateway;
Steps A 3, described local terminal gateway are received after the described response message, and described probe messages and response message thereof are carried out statistical analysis, draw the network quality of IP network between described local terminal gateway and the described opposite end gateway;
Steps A 4, when needs made a call, described local terminal gateway judged whether described network quality satisfies pre-provisioning request, if judged result is for being then initiate voip call;
Wherein in the voip call process, also include the following step:
Step B1, the RTP service quality that described local terminal gateway is monitored described IP network by recipient's report and the Sender Report of RTCP;
Step B2, if described local terminal gateway detects described RTP service quality when being lower than presetting first threshold, then the backstage is communicated with the PSTN circuit;
Step B3 if described local terminal gateway detects described RTP service quality when being lower than the second default threshold value, then enables the PSTN circuit that has been communicated with the former calling that continues, and stops former voip call simultaneously.
6. as the method for VoIP business is provided as described in the claim 5, it is characterized in that, in the voip call process, also include the following step:
Steps A 5, described local terminal gateway judge from sending described probe messages to the stand-by period of receiving its response message whether exceed the scheduled wait time, if judged result is for being then initiate the PSTN calling with the former voip call that continues.
7. the method for VoIP business is provided as described in arbitrary as claim 5 to 6, it is characterized in that described probe messages is the UDP message.
8. the method for VoIP business is provided as described in arbitrary as claim 5 to 6, it is characterized in that described response message is stamped timestamp.
9. method that the VoIP business is provided, it is characterized in that, in the voip call process, by the RTP service quality that recipient's report and the Sender Report of RTCP are monitored IP network, make the local terminal gateway whether to initiate PSTN and call out the former voip call that continues according to described RTP service quality decision;
Include the following step:
Step B1, the RTP service quality that the local terminal gateway is monitored IP network by recipient's report and the Sender Report of RTCP;
Step B2, if described local terminal gateway detects described RTP service quality when being lower than presetting first threshold, then the backstage is communicated with the PSTN circuit;
Step B3 if described local terminal gateway detects described RTP service quality when being lower than the second default threshold value, then enables the PSTN circuit that has been communicated with the former calling that continues, and stops former voip call simultaneously.
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| CN200710110622XA CN101056283B (en) | 2007-06-07 | 2007-06-07 | Voice gateway and method for providing VoIP service |
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| CN105611516A (en) * | 2016-03-10 | 2016-05-25 | 杭州腾展科技有限公司 | Telephone calling system and telephone calling method |
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| CN112565013B (en) * | 2020-12-04 | 2022-09-23 | 迪爱斯信息技术股份有限公司 | Voice communication method, switch, IP terminal and system using IP network |
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| CN101056283A (en) | 2007-10-17 |
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