CN102833436A - Method and system for achieving VOIP (Voice over Internet Phone) agency - Google Patents
Method and system for achieving VOIP (Voice over Internet Phone) agency Download PDFInfo
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Abstract
The invention provides a method for achieving a VOIP (Voice over Internet Phone) agency. The method is characterized by comprising the steps as follows: S1: starting S2: determining whether a client incoming call request is received through a server, if so, carrying out SIP (Session Initiation Protocol) treatment through the server, and sending SIP response, transferring to S5, otherwise, transferring to S3; S3: determining whether to enable the server to call out, if so, inputting a client number, and sending the SIP request to the number, and carrying out S4, otherwise, transferring to S2; S4: determining whether the number is responded, if so, transferring to S5, otherwise, determining that the call is failed, and transferring to S2; S5: establishing a communication linkage between the server and the client, and carrying out voice data treatment and playing; and S6: finishing the communication. According to the method, the SIP protocol is adopted to establish the communication linkage; the SDP (Service Discovery Protocol) is adopted to transmit the real-time voice data through the RTP (Real Time Protocol), and the quality of the real-time voice transmission is fed back with the combination with the RTCP (Real Time Control Protocol), so as to achieve the purpose of implementing the communication among users of set-top boxes within a local area network.
Description
Technical field
The present invention relates to the VOIP field, belong to the practical applications field, relate in particular to the method and the system that on STB, realize VOIP.
Background technology
VOIP (Voice over Internet Protocol) is exactly with simulated sound signal (Voice) digitlization in brief, on IP data network (IP Network), does real-time transmission with the form of data packet (Data Packet).The maximum advantage of VoIP is to adopt the environment of Internet and global ip interconnection widely, and, better service more than traditional business are provided.The business such as transmission voice, fax, video and data that VoIP is can be on IP network cheap are transmitted and the storage forwarding of various information etc. like unified message, virtual telephony, virtual speech/fax mailbox, directory enquiry service, Internet call center, Internet call manager, video conference, ecommerce, fax storage.VoIP (Voice over Internet Protocol) be a kind of be main with IP phone, and release the technology of corresponding value-added service.The maximum advantage of VoIP is to adopt the environment of Internet and global ip interconnection widely, and, better service more than traditional business are provided.Is VoIP relatively cheap why? Voip phone only is a kind of application on the Internet, and the networking telephone does not receive control.Therefore, in essence, voip phone and Email, instant messaging or webpage do not have any difference, and they all can transmit at the machinery compartment that has passed through Internet connection.These machines can be computers, and perhaps wireless device is such as mobile phone or hand-held device or the like.Why VoIP serves some will be collected money, is some free? The VoIP service not only can be linked up the voip user, and can converse with the telephone subscriber, such as the user who uses traditional fixed telephone network and wireless phone network.To this part conversation, VoIP service provider must give fixed telephone network operator and wireless telecommunications operator payment cost of the phone call.The charge of this part will forward on voip user's head.Conversation between the online voip user can be free.Using VoIP, what need you do? You need Internet connection.This can be the most basic dial-in service, perhaps better broadband service, and your network connection speed is fast more, and the speech quality of VoIP is just good more.For example, high-speed wideband connects can make you simultaneously make a phone call, and one side is surfed the web.The user can select a kind of VoIP software to be mounted on desktop computer or the notebook computer.Then, computer just can have been conversed on the net.If the user wants the home phone number of oneself is converted into the VoIP dial system, he needs the help of adapter.VoIP software can be pre-installed separately in the hardware device of a kind of " simulation telephony adapter " (analog telephone adapter) by name, and simulation telephony adapter mainly is installed between home phone number and the broadband modem.
Along with the continuous development of TV tech, digital television signal replaces the life that the conventional analogue signal gets into people gradually.The product that with the STB is receiving terminal for digital television has obtained using widely, but traditional STB can only be realized unidirectional digital TV broadcasting service, can not satisfy people's demand far away.At home, most of user has used CATV set-top-box, and very big user base is arranged.
Along with the transformation with bilateral network of carrying out of the integration of three networks, vast market is provided for this two-way services of VOIP.Based on network speech business is the trend of era development, and Application V OIP becomes a kind of trend on STB, is the inevitable outcome of the integration of three networks.The transformation of bilateral network is for DTV STB provides vast market and development space.DTV STB is just interactive towards high definition, intelligence is open and multi-functional direction develops.Along with the continuous propelling of the integration of three networks, the transmission of the across a network of integrated service will become possibility, make can communicate between plurality of devices such as mobile device, PC, television set.People can come call peer through different terminal whenever and wherever possible, and life also will become more convenient.VOIP is as a kind of important application on the network, and based on network speech business is the trend of era development, and the application of VOIP on STB becomes a kind of trend, is the inevitable outcome of the integration of three networks.
At present, the signaling protocol of main flow has the H.323 standard of ITU-T proposition and the Session Initiation Protocol that IETF proposes.H.323 standard is the aggregate of an agreement, has stipulated transferring voice, video and data on the packet network that no service quality guarantees, is applicable to any packet switching network.And Session Initiation Protocol is a separate protocol, uses with the SDP agreement is collaborative, and the attribute of conversation can be set, and realizes that conversation connects.
Summary of the invention
The present invention is in order to realize the VOIP user agent on the STB, the state of monitor user ' in real time.
In order to solve the problems of the technologies described above, the technical scheme that the present invention adopts provides a kind of VOIP of realization agency's method, comprises the steps:
S1: beginning
S2: server has judged whether client incoming call request,
If said server carries out the SIP Request Processing, and send sip response, change S5;
If not, then change step S3:
S3: server judges whether to need initiatively to breathe out,
If, then import client numbers, and this number is sent the SIP request, carry out S4;
If not, then change S2;
S4: judge whether said number responds,
If then change S5;
If not, then judge call failure, change S2;
S5: set up conversation between server and client and is connected execution language data process and broadcast;
S6: end of conversation.
Concrete comprising among the step S5: the collection of speech data, speech coding, RTP encapsulation, real-time Transmission, RTP unpack, tone decoding and speech play; Said speech data collection comprises: the function initialization; Open specified file, obtain the attribute of audio frequency, the state that reads voice data is set again through the resolution file head; Start the call back function reading of data; Output to products for further processing in the opposition, every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, the speech data finishing collecting is described.The computing formula that at every turn reads the voice byte number is: Byte=2 * (Δ
tG.711, the said encoded speech data of * f * count)/1000 is through encoding; Carry out the analysis of speech data, comprise voice quality, speech encoding rate, algorithm complex, encoding time delay and the compatibility of speech data; Said speech play is play through the RCA COBBAIF connection television set of STB.
Said real-time Transmission, the concrete transmission that comprises speech data and the reception of speech data;
The process of the transmission of speech data:
S611: judge that whether first RTP sends startup, if first timestamp that RTP sends just is set;
S612: the timestamp of the current transmission of RTP is set, upgrades the sequence number of RTP and the size of data;
S613: judge whether to send the RTP bag, if then call the packet that transmission interface sends speech data;
Among the step S6, the receiving course of speech data:
S621: upgrade time of reception, call receiving interface;
S622: the type of judgment data bag, add the reception formation to;
S623: delay jitter is handled, the timestamp of the current transmission of RTP of the bag that Updates Information;
S624: the packet to receiving is handled.
Step S1 accomplishes following operation through SIP and SDP agreement:
S11: system initialization;
S12: parameter configuration and registration;
S13: create and detect thread.
In order to solve the problems of the technologies described above, the technical scheme that the present invention adopts provides a kind of VOIP of realization agency's system, comprising: first judging unit, second judging unit, the 3rd judging unit and processing unit; First judging unit is used to judge whether client incoming call request, if then carry out the SIP Request Processing, and send sip response; If not, then send no incoming call solicited message; Second judging unit is used to judge whether initiatively exhalation of needs, if then import client numbers, and send the SIP request to said number; The 3rd judging unit is used to judge whether client responds, if then send information to said processing unit; If not, then judge call failure, and make the first judging unit work; Processing unit is used to make and sets up conversation between server and client and is connected execution language data process and broadcast.
Said processing unit comprises: speech data collection module, speech coding module, RTP package module, real-time Transmission module, RTP parse module, tone decoding module and speech play module; Said speech data collection module is used for the function initialization; Open specified file, obtain the attribute of audio frequency, the state that reads voice data is set again through the resolution file head; Start the call back function reading of data; Output to products for further processing in the opposition, every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, the speech data finishing collecting is described.The computing formula that at every turn reads the voice byte number is: Byte=2 * (Δ
t* f * count)/1000 said encoded speech data module is used for carrying out the analysis of speech data through G.711 encoding, and comprises voice quality, speech encoding rate, algorithm complex, encoding time delay and the compatibility of speech data; Said speech play module is used for playing through the RCA COBBAIF connection television set of STB
Said real-time Transmission module comprises speech data sending module speech data receiver module; Said speech data sending module is used to judge that whether first RTP sends startup, if first timestamp that RTP sends just is set; And the timestamp of the current transmission of RTP is set, upgrades the sequence number of RTP and the size of data; Judge whether to send the RTP bag, if then call the packet that transmission interface sends speech data; Said speech data receiver module is used to upgrade time of reception, calls receiving interface; The type of judgment data bag is added the reception formation to; The packet that receives is carried out delay jitter handle, the timestamp of the bag that Updates Information; Packet to receiving is handled.
Said system also comprises signaling process and control unit, and said signaling process and control unit are used for accomplishing system initialization through SIP and SDP agreement, parameter configuration and registration and the establishment that detects thread.
The present invention also provides a kind of VOIP of realization agency's system; Said system comprises PC equipment that is used as server and the STB that is used as client; Said PC equipment realizes that with STB conversation is connected through Session Initiation Protocol, SDP agreement, and any described method of said PC equipment and set-top box application claim 1 to 4 is carried out communication.
The invention has the beneficial effects as follows:
Through PC equipment and STB, using SIP agreement, SDP agreement are set up conversation and are connected, and adopt described realization VOIP agency's method; When entering into the VOIP application interface, will create a thread and come monitoring in real time, judge type of message through the message that parsing receives; Carry out corresponding the processing simultaneously; After both call sides is through foundation conversation connection end to end, then carry out collection, voice real-time Transmission, speech processes and the speech play of speech data, technical scheme using SIP agreement of the present invention is set up conversation and is connected; Utilize the SDP agreement to consult the attribute of voice call; Choose G.711 that code decode algorithm carries out encoding compression to speech data, use Real-time Transport Protocol to transmit the real-time voice data, and combine rtcp protocol to feed back the quality of real-time voice transmission.It can realize set-top box users registration, call out functions such as control, voice call, thereby reach the purpose of in local area network (LAN), conversing between the set-top box users.
Description of drawings
Fig. 1 is the flow chart of realization VOIP Proxy Method of the present invention;
Fig. 2 is that conversation provided by the invention connects certain embodiment flow chart of setting up;
Fig. 3 is the structured flowchart of realization VOIP agency plant provided by the invention.
Embodiment
By specifying technology contents of the present invention, structural feature, realized purpose and effect, give explanation below in conjunction with execution mode and conjunction with figs. are detailed.
Realization VOIP agency's provided by the invention method; This method is applied to realize in VOIP agency's the system; Said system comprises PC equipment that is used as server and the STB that is used as client, and said PC equipment realizes that with STB conversation is connected through Session Initiation Protocol, SDP agreement.Concrete, consult Fig. 1, realize that VOIP agency's method comprises the steps:
S1: beginning; In the unlatching of VOIP agency plant, the VOIP agency plant is on the frame structure of VOIP software, through signaling process and control unit; Using SIP, SDP agreement, the establishment of between server and client, accomplishing system initialization, parameter configuration and registration and detecting thread; The establishment of said detection thread is in order to realize detecting the follow-up sip message generation that whether has; Said sip message comprises two kinds of request message and response messages; Mainly contain six kinds of REGISTER, INVITE, ACK, OPTION, BYE, CANCEL, and sip response message there is 1XX-6XX; In the process of parameter configuration and registration; Concrete needs judge whether that whether overtime registration and hour of log-on etc.; The purpose of registration mainly is in order to guarantee that the user can be at the enterprising line item of server; Guarantee that other users can find this user, the detection of two registration timeouts is in order to prevent withdrawing from unusually of user, to cause the generation of situation such as the user can't nullify.
S2: server has judged whether client incoming call request, if said server carries out the SIP Request Processing, and sends sip response, changes S5; If not, then change step S3; In this process, through first judging unit, judged whether client incoming call request, when first judgment unit judges does not have client incoming call request, then send no incoming call solicited message to second judging unit.
S3: server judges whether to need initiatively to breathe out, if, then import client numbers, and this number is sent the SIP request, carry out S4; If not, then change S2; After second judging unit receives the no incoming call request message of first judging unit, judge then whether said server needs initiatively to breathe out, if then import client numbers, and send the SIP request to said number.
S4: judge whether said number responds, if then change S5; If not, then judge call failure, change S2; The 3rd judging unit is through judging whether called number is corresponding; If judge called number response, execution in step S5 then is if judge the called number no response; Then return step S2, and make first judging unit of said server restart to have judged whether client incoming call request.
S5: set up conversation between server and client and is connected execution language data process and broadcast; Described language data process with play concrete comprising: the collection of speech data, speech coding, RTP encapsulation, real-time Transmission, RTP unpack, tone decoding and speech play;
Said speech data collection comprises: the function initialization; Open specified file, obtain the attribute of audio frequency, the state that reads voice data is set again through the resolution file head; Start the call back function reading of data; Output to products for further processing in the formation, every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, the speech data finishing collecting is described.The computing formula that at every turn reads the voice byte number is: Byte=2 * (Δ
t* f * count)/1000.The collection of speech data is accomplished through speech data collection module.
The encoded speech data module is accomplished encoded speech data, and concrete is through G.711 encoding, carry out the analysis of speech data, comprising voice quality, speech encoding rate, algorithm complex, encoding time delay and the compatibility of speech data;
Said speech play is play through the RCA COBBAIF connection television set of STB.
In step S5, between server and client, set up good conversation and be connected, carry out in language data process and the playing process; At first will carry out the collection of speech data, the collection of speech data is accomplished through speech data collection module, limited by the STB external interface; The collection of speech data is to read in the audio file that adopts from USB flash disk, and detailed process is the function initialization; In generation, opened specified file, obtains the attribute of audio frequency through the resolution file head, and the attribute of audio frequency has mainly comprised information such as channel number, sample rate; Thereby can obtain total amount of data, read the state that voice data is set behind the audio attribute, comprise information such as the stopping of voice data, beginning and end; After having obtained the relevant information of voice data, start the call back function reading of data, output to products for further processing in the formation; Every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, explain that speech data reads to finish.The computing formula that at every turn reads the voice byte number is:
Byte=2×(Δ
t×f×count)/1000
Byte is the byte number that reads, Δ
tBe the interval of reading of data, f is a sample rate, and count is a channel number.
After the voice messaging finishing collecting; Carry out encoded speech data; Speech coding is the important component part of speech processes, is one of key technology of network voice transmission, is the indispensable function of voice terminal; The algorithm selection of encoding and decoding needs to consider the factor of a lot of aspects, and it has directly determined the performance of encoding and decoding speech.What the present invention used is the G711 code decode algorithm, mainly is because the realization of G711 code decode algorithm is fairly simple, belongs to waveform coding; Faithful to primary voice data; Voice quality is relatively good, and the disposal ability of the set-top-box chip that this time uses is limited, though the network bandwidth that use G711 code decode algorithm takies is than higher; But the realization in local area network (LAN), bandwidth are enough to satisfy the needs of voice output.
Carry out the concrete transmission that comprises speech data of voice data transmission and the reception of speech data,
The process of transmitting of speech data comprises the steps:
S611: judge that whether first RTP sends startup, if first timestamp that RTP sends just is set;
S612: the timestamp of the current transmission of RTP is set, upgrades the sequence number of RTP and the size of data;
S613: judge whether to send the RTP bag, if then call the packet that transmission interface sends speech data.
In the process of sending, to judge that at first whether first RTP sends startup, if wine is provided with first timestamp of transmission; The initial value of transmitting time is set, is equivalent to the initialization of carrying out system constantly at this, the timestamp of RTP timestamp and current transmission is set again; The timestamp of RTP is when unpacking for RTP; Data continuously with synchronously, the timestamp of current transmission is in order to stab update time, and upgrades the size of sequence number and data; Wherein the sequence number of RTP is in order to see if there is the phenomenon of data packet loss, and the size of data is the total lengths that send data in order to calculate.RTP bag through special sends function, has judged whether that the RTP bag sends, and is true if send the return value of function, and then representative need be carried out the transmission of RTP, and calls the transmission that transmission interface carries out the RTP bag.
The receiving course of speech data comprises the steps:
S621: upgrade time of reception, call receiving interface;
S622: the type of judgment data bag, add the reception formation to;
S623: delay jitter is handled, the timestamp of the current transmission of RTP of the bag that Updates Information;
S624: the packet to receiving is handled.
The process that RTP receives speech data is that RTP timestamp and the sequence number when sending unpacks speech data, at first upgrades time of reception, calls receiving interface; The type of judgment data bag then, described packet comprises incident bag and VoP, and described packet is added in the corresponding reception formation; Then carry out delay jitter and handle, the timestamp of the bag that Updates Information is owing to the reason of network; The data of receiving maybe be in proper order inconsistent, handles through delay jitter, makes data to unpack by the order of sending; The timestamp here be stream timestamp, for data rearrangement used.Carry out the reception of RTP bag at last and handle, and issue the upper strata and handle, the upper strata carries out that RTP unpacks and tone decoding.Above-mentioned RTP sends with receiving to handle through the ORTP code of increasing income and realizes.
At last speech data is play, the broadcast of speech data is to play through the RCA COBBAIF connection television set of STB, and concrete play parameter has comprised the bit wide of sampling, big small end storage, PCM sample rate and the channel number of data.General STB all can be supported the broadcast of pure PCM data, and an interface function of playing the PCM data can be provided.The process of speech play realizes through two threads.One is the thread that receives speech data, and speech data is put into broadcast area from buffering area.One is to play thread, and the data in the broadcast area are play.Between two threads is realization with mutual exclusion through a semaphore synchronously.
S6: end of conversation.
Consult Fig. 2, represented to set up between PC equipment of the present invention and the STB flow chart of the embodiment that conversation is connected among the figure.Wherein, Described PC equipment has the function of VOIP service, and STB is realized the final network and the conversation procedure of television set through connecting television set, and wherein user A desires to converse with user B; Equipment such as user A application television machine; User B Using P C equipment, said PC equipment possess the function of supporting VOIP, as the VOIP server end.The television set of user A is connected with STB, thereby finally is to communicate with server end through STB to be connected.
After the STB calling party B; The call request of VOIP acting server responder top box, through the address of VOIP acting server location, inquiring user B, and the address of response user B; Acting server is acted on behalf of said STB calling party B; The message of user B response agent server, and finally be sent to STB to corresponding signal, accomplish the conversation establishment of connection.User A realizes that through Session Initiation Protocol, SDP agreement conversation is connected with user B, and said PC equipment and the above-described method of set-top box application are carried out communication, and the completion server is connected with client.
Consult Fig. 3, the present invention also provides a kind of VOIP of realization agency's system, comprising: first judging unit, second judging unit, the 3rd judging unit and processing unit;
First judging unit is used to judge whether client incoming call request, if then carry out the SIP Request Processing, and send sip response; If not, then send no incoming call solicited message;
Second judging unit is used to judge whether initiatively exhalation of needs, if then import client numbers, and send the SIP request to said number;
The 3rd judging unit is used to judge whether client responds, if then send information to said processing unit; If not, then judge call failure, and make the first judging unit work;
Processing unit is used to make and sets up conversation between server and client and is connected execution language data process and broadcast.
Said processing unit comprises: speech data collection module, speech coding module, RTP package module, real-time Transmission module, RTP parse module, tone decoding module and speech play module;
Said speech data collection module is used for the function initialization; Open specified file, obtain the attribute of audio frequency, the state that reads voice data is set again through the resolution file head; Start the call back function reading of data; Output to products for further processing in the opposition, every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, the speech data finishing collecting is described.The computing formula that at every turn reads the voice byte number is: Byte=2 * (Δ
t* f * count)/1000
Said encoded speech data module is used for carrying out the analysis of speech data through G.711 encoding, and comprises voice quality, speech encoding rate, algorithm complex, encoding time delay and the compatibility of speech data;
Said speech play module is used for playing through the RCA COBBAIF connection television set of STB
Said real-time Transmission module comprises speech data sending module speech data receiver module;
Said speech data sending module is used to judge that whether first RTP sends startup, if first timestamp that RTP sends just is set; And the timestamp of the current transmission of RTP is set, upgrades the sequence number of RTP and the size of data; Judge whether to send the RTP bag, if then call the packet that transmission interface sends speech data;
Said speech data receiver module is used to upgrade time of reception, calls receiving interface; The type of judgment data bag is added the reception formation to; The packet that receives is carried out delay jitter handle, the timestamp of the bag that Updates Information; Packet to receiving is handled.
Said system also comprises signaling process and control unit, and said signaling process and control unit are used for accomplishing system initialization through SIP and SDP agreement, parameter configuration and registration and the establishment that detects thread.
Said system also comprises human-computer interaction module; One output of human-computer interaction module is connected with control unit with signaling process; The other end is connected with processing unit; Described human-computer interaction module comprises voice receiving equipment (microphone), voice receiving equipment (loudspeaker), PC keyboard etc., and user A is through after the incoming call user B, opening whole system through above-described each unit on the PC keyboard; And then the conversation of described processing unit processes user A, user B connects signal, thereby reaches the purpose that user A and user B communicate with each other through television set and PC end subscriber.
A kind of system that realizes the VOIP agency; Said system comprises PC equipment that is used as server and the STB that is used as client; Said PC equipment realizes that with STB conversation is connected through Session Initiation Protocol, SDP agreement; Said PC equipment and the above-described method of set-top box application are carried out communication, and the completion server is connected with client.
The above is merely embodiments of the invention; Be not so limit claim of the present invention; Every equivalent structure or equivalent flow process conversion that utilizes specification of the present invention and accompanying drawing content to be done; Or directly or indirectly be used in other relevant technical fields, all in like manner be included in the scope of patent protection of the present invention.
Claims (9)
1. a method that realizes the VOIP agency is characterized in that, comprises the steps:
S1: beginning
S2: server has judged whether client incoming call request,
If said server carries out the SIP Request Processing, and send sip response, change S5;
If not, then change step S3:
S3: server judges whether to need initiatively to breathe out,
If, then import client numbers, and this number is sent the SIP request, carry out S4;
If not, then change S2;
S4: judge whether said number responds,
If then change S5;
If not, then judge call failure, change S2;
S5: set up conversation between server and client and is connected execution language data process and broadcast;
S6: end of conversation.
2. realization according to claim 1 VOIP agency's method is characterized in that concrete comprising among the step S5: the collection of speech data, speech coding, RTP encapsulation, real-time Transmission, RTP unpack, tone decoding and speech play;
Said speech data collection comprises: the function initialization; Open specified file, obtain the attribute of audio frequency, the state that reads voice data is set again through the resolution file head; Start the call back function reading of data; Output to products for further processing in the opposition, every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, the speech data finishing collecting is described.The computing formula that at every turn reads the voice byte number is: Byte=2 * (Δ
t* f * count)/1000
Said encoded speech data is carried out the analysis of speech data through G.711 encoding, and comprises voice quality, speech encoding rate, algorithm complex, encoding time delay and the compatibility of speech data;
Said speech play is play through the RCA COBBAIF connection television set of STB.
3. realization VOIP agency's according to claim 2 method is characterized in that said real-time Transmission, the concrete transmission that comprises speech data and the reception of speech data;
The process of the transmission of speech data:
S611: judge that whether first RTP sends startup, if first timestamp that RTP sends just is set;
S612: the timestamp of the current transmission of RTP is set, upgrades the sequence number of RTP and the size of data;
S613: judge whether to send the RTP bag, if then call the packet that transmission interface sends speech data;
Among the step S6, the receiving course of speech data:
S621: upgrade time of reception, call receiving interface;
S622: the type of judgment data bag, add the reception formation to;
S623: delay jitter is handled, the timestamp of the current transmission of RTP of the bag that Updates Information;
S624: the packet to receiving is handled.
4. realization VOIP agency's according to claim 1 method is characterized in that, step S1 accomplishes following operation through SIP and SDP agreement:
S11: system initialization;
S12: parameter configuration and registration;
S13: create and detect thread.
5. a system that realizes the VOIP agency is characterized in that, comprising: first judging unit, second judging unit, the 3rd judging unit and processing unit;
First judging unit is used to judge whether client incoming call request, if then carry out the SIP Request Processing, and send sip response; If not, then send no incoming call solicited message;
Second judging unit is used to judge whether initiatively exhalation of needs, if then import client numbers, and send the SIP request to said number;
The 3rd judging unit is used to judge whether client responds, if then send information to said processing unit; If not, then judge call failure, and make the first judging unit work;
Processing unit is used to make and sets up conversation between server and client and is connected execution language data process and broadcast.
6. realization VOIP agency's according to claim 5 system; It is characterized in that said processing unit comprises: speech data collection module, speech coding module, RTP package module, real-time Transmission module, RTP parse module, tone decoding module and speech play module;
Said speech data collection module is used for the function initialization; Open specified file, obtain the attribute of audio frequency, the state that reads voice data is set again through the resolution file head; Start the call back function reading of data; Output to products for further processing in the opposition, every 20ms gets once, and the state of reading of data is set after having read.At last, when the byte number that reads is less than the byte number of regulation, the speech data finishing collecting is described.The computing formula that at every turn reads the voice byte number is: Byte=2 * (Δ
t* f * count)/1000
Said encoded speech data module is used for carrying out the analysis of speech data through G.711 encoding, and comprises voice quality, speech encoding rate, algorithm complex, encoding time delay and the compatibility of speech data;
Said speech play module is used for playing through the RCA COBBAIF connection television set of STB.
7. realization VOIP agency's according to claim 6 system is characterized in that said real-time Transmission module comprises speech data sending module and speech data receiver module;
Said speech data sending module is used to judge that whether first RTP sends startup, if first timestamp that RTP sends just is set; And the timestamp of the current transmission of RTP is set, upgrades the sequence number of RTP and the size of data; Judge whether to send the RTP bag, if then call the packet that transmission interface sends speech data;
Said speech data receiver module is used to upgrade time of reception, calls receiving interface; The type of judgment data bag is added the reception formation to; The packet that receives is carried out delay jitter handle, the timestamp of the bag that Updates Information; Packet to receiving is handled.
8. realization VOIP agency's according to claim 5 system; It is characterized in that: said system also comprises signaling process and control unit; Said signaling process and control unit are used for accomplishing system initialization through SIP and SDP agreement, parameter configuration and registration and the establishment that detects thread.
9. system that realizes VOIP agency; It is characterized in that; Said system comprises PC equipment that is used as server and the STB that is used as client; Said PC equipment realizes that with STB conversation is connected through Session Initiation Protocol, SDP agreement, and any described method of said PC equipment and set-top box application claim 1 to 4 is carried out communication.
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