System and method for supporting VOIP and CS telephone
AssigneeZte Corporation
InventorQiang Wang
Life of this patent
Estimating the term…
Abstract
The present invention discloses a system for supporting Voice over Internet Protocol (VOIP) and Circuit Switch (CS) telephone. The system includes a terminal gateway device, a wireless broadband module, a Subscriber Line Interface Circuit (SLIC), a SWITCH and a telephone. The present invention also discloses a method for supporting VOIP and CS telephone. By adopting the system and the method of the present invention, a subscriber can enjoy different voice services on the same telephone according to different service scenes, and a mobile operator can deploy a corresponding service in accordance with specific conditions; the present invention enhances the value-adding capability and saves communication cost.
Claims8
A system for supporting Voice over Internet Protocol (VOIP) and Circuit Switch (CS) telephone, comprising a terminal gateway device, a wireless broadband module and a Subscriber Line Interface Circuit (SLIC), wherein
- the wireless broadband module is used for realizing network registration of a VOIP call, connection to a Wide Area Network (WAN) and establishment of a CS call and completing transmission of a voice code stream between the terminal gateway device and a wireless broadband network;
- the terminal gateway device is used for determining whether a currently initiated call is a VOIP call or a CS call according to a received signal;
- if the currently initiated call is a VOIP call, establishing a Session Initiation Protocol (SIP) connection via the wireless broadband module, converting a Pulse Code Modulation (PCM) line signal into a VOIP voice code stream, and sending the converted VOIP voice code stream to the wireless broadband network via the wireless broadband module or converting a VOIP voice code stream from the wireless broadband module into a PCM line signal;
- if the currently initiated call is a CS call, establishing a CS connection via the wireless broadband module, converting a PCM line signal into a CS voice code stream, and sending the converted CS voice code stream to the wireless broadband network via the wireless broadband module or converting a CS voice code stream from the wireless broadband module into a PCM line signal;
- and
- the SLIC is used for bidirectional transmission of a Serial Peripheral Interface (SPI) signal between a telephone and the terminal gateway device as well as bidirectional transmission of a PCM line signal between a telephone and the terminal gateway device;
- wherein the terminal gateway device comprises an SLIC drive module, a voice Digital Signal Processing (DSP) module, a Real-Time Transmission Protocol (RTP)/RTP Control Protocol (RTCP) module, an SIP module and a voice application module, wherein
- the SLIC drive module is used for detecting an SPI signal coming from the SLIC, determining whether the currently initiated call is a VOIP call or a CS call, encoding a PCM line signal into a VOIP or CS voice code stream corresponding to the type of the call, sending the encoded voice code stream to the voice DSP module, and decoding a VOIP voice code stream or CS voice code stream from the voice DSP module to a PCM line signal;
- the voice DSP module is used for compressing and encapsulating an encoded voice stream to obtain a voice data packet, sending the voice data packet to the RTP/RTCP module in the case of a VOIP call or to the voice application module in the case of a CS call, and unpacking and decompressing a voice data packet from the RTP/RTCP module or the voice application module to obtain a VOIP voice code stream or CS voice code stream;
- the RTP/RTCP module is used for real-time transmission control of a voice data packet over an IP network for a VOIP call and is mainly used for sending a voice data packet from the voice DSP module to the wireless broadband module via the voice application module and further to the IP network via the wireless broadband module, and sending a voice data packet received by the voice application module to the voice DSP module;
- the SIP module is used for creating, modifying and releasing a VOIP session;
- and
- the voice application module is used for identifying a voice code stream coming from the wireless broadband module, determining whether a currently initiated call is a VOIP call or CS call;
- if the currently initiated call is a VOIP call, establishing a connection between the terminal gateway device and the wireless broadband module through a Point-to-Point Protocol (PPP) and sending a received voice data packet to the RTP/RTCP module;
- if the currently initiated call is a CS call, sending a received voice code stream to the voice DSP module.
A system for supporting Voice over Internet Protocol (VOIP) and Circuit Switch (CS) telephone, comprising a terminal gateway device, a wireless broadband module, a Subscriber Line Interface Circuit (SLIC) and a SWITCH, wherein
- the wireless broadband module is used for realizing network registration of a VOIP call, connection to a WAN and establishment of a CS call, and completing transmission of a voice code stream between the terminal gateway device and a wireless broadband network;
- further for establishing a CS connection if a CS voice code stream is transmitted through a Pulse Code Modulation (PCM) line from the SLIC to the wireless broadband module, converting a PCM line signal into a CS voice code stream and sending the converted CS voice code stream to the wireless broadband network, and converting a CS voice code stream from the wireless broadband network into a PCM line signal;
- the terminal gateway device is used for determining whether a currently initiated call is a VOIP call or a CS call according to a received signal;
- if the currently initiated call is a VOIP call, establishing a Session Initiation Protocol (SIP) connection via the wireless broadband module, converting a PCM line signal into a VOIP voice code stream, and sending the converted VOIP voice code stream to the wireless broadband network via the wireless broadband module or converting a VOIP voice code stream from the wireless broadband module into a PCM line signal;
- if the currently initiated call is a CS call, establishing a CS connection via the wireless broadband module, converting a PCM line signal into a CS voice code stream, and sending the converted CS voice code stream to the wireless broadband network via the wireless broadband module or converting a CS voice code stream from the wireless broadband module into a PCM line signal;
- the terminal gateway device is further used for determining whether a PCM line for transmitting a CS voice code stream is from the SLIC to the terminal gateway device or from the SLIC to the wireless broadband module, and controlling the SWITCH to establish a corresponding PCM line for transmitting the CS voice code stream;
- the SLIC is used for bidirectional transmission of a Serial Peripheral Interface (SPI) signal between a telephone and the terminal gateway device as well as bidirectional transmission of a PCM line signal between a telephone and the SWITCH;
- and
- the SWITCH is used for transmission of a VOIP voice code stream between the terminal gateway device and the SLIC, and establishment of a PCM line from the SLIC to the terminal gateway device or to the wireless broadband module for transmitting a CS voice code stream;
- wherein the terminal gateway device comprises an SLIC drive module, a voice Digital Signal Processing (DSP) module, a Real-Time Transmission Protocol (RTP)/RTP Control Protocol (RTCP) module, an SIP module, a voice application module and a Web subscriber management interface module, wherein
- the SLIC drive module is used for detecting an SPI signal coming from the SLIC, determining whether a currently initiated call is a VOIP call or a CS call, encoding a PCM line signal to a VOIP or CS voice code stream corresponding to the type of the call, sending the encoded voice code stream to the voice DSP module, and decoding a VOIP or CS voice code stream from the voice DSP module to a PCM line signal;
- the voice DSP module is used for compressing and encapsulating an encoded voice code stream to obtain a voice data packet, sending the voice data packet to the RTP/RTCP module in the case of a VOIP call or to the voice application module in the case of a CS call, and unpacking and decompressing a voice data packet from the RTP/RTCP module or the voice application module to obtain a VOIP voice code stream or CS voice code stream;
- the RTP/RTCP module is used for real-time transmission control of a voice data packet over an IP network for a VOIP call and is mainly for sending a voice data packet from the voice DSP module to the wireless broadband module via the voice application module and further to the IP network via the wireless broadband module, and sending a voice data packet received by the voice application module to the voice DSP module;
- the SIP module is used for creating, modifying and releasing a VOIP session;
- the voice application module is used for identifying a voice code stream coming from the wireless broadband module, determining whether a currently initiated call is a VOIP call or CS call;
- if the currently initiated call is a VOIP call, establishing a connection between the terminal gateway device and the wireless broadband module through a PPP, and sending a received voice data packet to the RTP/RTCP module;
- if the currently initiated call is a CS call, sending a received voice code stream to the voice DSP module;
- and
- the Web subscriber management interface module is used for controlling, if the SLIC drive module determines the currently initiated call is a CS call, the SWITCH to establish a PCM line from the SLIC to the terminal gateway device or from the SLIC to the wireless broadband module for transmitting a CS voice code stream.
A method for supporting Voice over Internet Protocol (VOIP) and Circuit Switch (CS) telephone, comprising a calling process and a called process, wherein
- the calling process mainly comprises:
- a user picks up a telephone, an off-hook signal is sent to a terminal gateway device via a Serial Peripheral Interface (SPI), the terminal gateway device plays an off-hook tone and sends the off-hook tone to the telephone via a Subscriber Line Interface Circuit (SLIC), and the off-hook tone is played to the user via the telephone;
- the user dials a VOIP or CS number in different ways, and the terminal gateway device identifies a dialing signaling;
- the user establishes a VOIP or CS communication connection with a called party, a Pulse Code Modulation (PCM) line signal of each of the communications is converted into a corresponding voice code stream and then the corresponding voice code stream is transmitted to the called party, and a voice code stream of the called party is converted into a PCM line signal and then the PCM line signal is transmitted to the caller;
- and
- the user hangs up, the terminal gateway device detects a hand-up button and releases the communication connection;
- and the called process comprises:
- the terminal gateway device receives a call request from the caller and determines whether the received call request is a VOIP request or a CS request;
- the called party establishes a VOIP or CS session connection with the caller, and the SLIC transmits ring information and a telephone number to the telephone;
- the called party picks up a telephone, a VOIP or CS voice code stream is converted into a PCM line signal and then the PCM line signal is transmitted to the SLIC via a PCM line and finally to the user via the telephone, and a PCM line signal of the called party is converted into a VOIP or CS voice code stream and then the VOIP or CS voice code stream is transmitted to the caller;
- and
- the called party hangs up, and a wireless broadband module sends a received link removal request to the terminal gateway device, the terminal gateway device releases the session, and the whole call process is ended;
- wherein the process that the user establishes a VOIP or CS communication connection with a called party and voice code streams are transmitted between the user and the called party during the calling process comprises:
- the terminal gateway device sends a dial-up connection request to the called party via the wireless broadband module, a communication is established if the called party accepts the request;
- if the communication established is a VOIP communication, a PCM line signal is converted into a VOIP voice data packet by the terminal gateway device and then the VOIP voice data packet is transmitted to the called party;
- and a VOIP voice data packet from the called party is sent to the terminal gateway device via the wireless broadband module and converted into a PCM line signal and then the PCM line signal is transmitted to the caller;
- and
- if the communication established is a CS communication, if a CS voice code stream is transmitted through a PCM line from the SLIC to the terminal gateway device, a PCM line signal is converted into a CS voice code stream via the terminal gateway device and the converted CS voice code stream is transmitted to the called party via the wireless broadband module;
- and a CS voice code stream from the called party is decoded to a PCM line signal by the terminal gateway device and then the PCM line signal is transmitted to the caller;
- if a CS voice code stream is transmitted through a PCM line from the SLIC to the wireless broadband module, a PCM line signal is converted into a CS voice code stream via the wireless broadband module and the converted CS voice code stream is transmitted to the called party;
- and a CS voice code stream of the called party is converted into a PCM line signal and then the PCM line signal is transmitted to the caller.
Dependencies are inferred from the claim text.