Abstract:
An IP telephony apparatus provides simultaneous SIP communication for multiple IP phones and a method is proposed for the same. The IP telephony apparatus comprises a network connection port connected to a network; a plurality of telephone connection ports connected to a plurality of telephone sets; a voice codec unit connected to the telephone connection ports and used for converting a voice signal to a digital voice packet and for converting a digital voice packet to a voice signal; and an IP telephony allocation unit connected to the network connection port and the plurality of telephone connection ports. The IP telephony allocation unit creates an SIP control block containing an SIP call-leg for a local SIP number, and allocates the SIP control block to a destination telephone set according to the SIP call-leg, whereby the telephone sets have bi-directional digital voice packet transmission with a remote SIP number.

Description:
BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention relates to an IP telephony apparatus providing simultaneous communication for multiple IP phones and a method for the same, especially to an IP telephony apparatus providing simultaneous SIP communication for multiple IP phones by using only one SIP number and a method for the same. 
   2. Description of Prior Art 
   The progress of Internet technology provides innovative ideas and new services. For example, the VoIP (Voice over Internet Protocol) technique provides telephony communication through IP network and the expensive long-distance call fee can be saved. More particularly, the VoIP technique provides PC-to-PC telephony communication through IP network, PC-to-Phone telephony communication through PBX (private branch box), Phone-to-Phone telephony communication through ISP gateway and Device-to-Device telephony communication through IP Phones. 
   The VoIP technique provides suitable telephony signal and voice transmission for conveying phone call through IP network. The VoIP technique sends telephony signal with specific protocol to represent user status and to establish communication for user. Once the communication is established for user, the voice is compressed and digitalized to transmit in the form of digital signal. 
   The conventional telephony signals such as dialing signals, ringing signal, and busy signal are converted into data packets according to VoIP protocol and the data packets a resent to remote user through IP network. The data packets are then converted to analog telephony signal by remote IAD or ATA for the operation of remote telephone set. 
   After the connection is established, the analog voice is sent to a local router through a telephone set, a fax or a PBX. The analog voice is compressed and digitalized into data packets and the data packets are sent to remote router through IP network. The data packets are converted to analog voice signal by the remote router and then sent to remote user through a telephone set, a fax or a PBX. The user can make cheaper long-distance phone call using VoIP technique through omnipresent IP network instead of the conventional PSTN system. 
   The present VoIP technique is regulated by ITU (International Telecommunication Union) and the earlier protocol such as H323/H248 are defined for LAN rather than the open environment of Internet. Therefore the H323/H248 protocol has limited application and complicated conversion for PSTN system. A new protocol, namely SIP (Session Initiation Protocol) protocol, is defined by IETF (Internet Engineering Task Force) to fully exploit the Internet service and provide better integration of Internet and PSTN system. 
   The SIP protocol belongs to the application layer protocol in the seven-layer architecture of the OSI (Open System Interface) and is resemblant to the Client-Server structure in HTTP protocol. Therefore, the SIP protocol can utilize existing HTTP packet structure for sending command and data and can be adapted for data transmission in WAN. 
   In present VoIP telephone system, a UA ((User Agent) such as a VoIP gateway is need to install at user side and at least one call server should be installed at VoIP agent. Moreover, the VoIP user needs to register an SIP VoIP number to the VoIP agent. Therefore, other VoIP user can call him through the SIP VoIP number. 
   However, the nowadays VoIP gateway generally uses VoIP H323/H248 protocol and a VoIP telephone set such as a USB telephone set is required. When the VoIP gateway is connected to multiple VoIP telephone sets, each of the VoIP telephone set needs a unique SIP VoIP number to prevent blocked call. Therefore, the conventional VoIP gateway needs at least two SIP VoIP numbers for two VoIP telephone sets to prevent blocked call. 
   SUMMARY OF THE INVENTION 
   It is an object of the present invention to provide an IP telephony apparatus providing simultaneous SIP communication for multiple IP phones by using only one SIP number and a method for the same, wherein a call-leg is used to designate different SIP. The call-leg is established only when an SIP number is dialed in or out. The SIP call-leg can be dynamically created with no limitation on number thereof, whereby simultaneous SIP communication for multiple IP phones can be provided without blocked problem. 
   To achieve the above object, the present invention provides an IP telephony apparatus provides simultaneous SIP communication for multiple IP phones. The IP telephony apparatus comprises a network connection port connected to a network; a plurality of telephone connection ports connected to a plurality of telephone sets; a voice codec unit connected to the telephone connection ports and used for converting a voice signal to a digital voice packet and for converting a digital voice packet to a voice signal; and an IP telephony allocation unit connected to the network connection port and the plurality of telephone connection ports. The IP telephony allocation unit creates an SIP control block containing an SIP call-leg for a local SIP number, and allocates the SIP control block to a destination telephone set according to the SIP call-leg, whereby the telephone sets have bi-directional digital voice packet transmission with a remote SIP number. 
   To achieve the above object, the present invention provides a method providing simultaneous phone dialing for multiple IP phones. The method comprises following steps. A remote SIP number is dialed with at least one telephone set. A local SIP number is found for communication with the remote SIP number by a mapped policy means. At least one SIP control block containing an SIP call-leg is created for the local SIP number. The SIP control block is allocated to the dialing telephone set according to the SIP call-leg and to establish a bi-directional digital voice packet transmission between the dialing telephone set and the remote SIP number. When one telephone is using, other telephones can establish new SIP control block with SIP call-leg to dial out and will not suffer to the blocked problem. 
   To achieve the above object, the present invention provides a method providing simultaneous call receiving for multiple IP phones. The method comprises following steps. A phone call from a remote SIP number for calling a local SIP number is received. At least one SIP control block containing an SIP call-leg is created for a telephone group containing a plurality of telephone sets for making a ringing for all telephone sets in the telephone group. When a telephone set of the telephone group is off hook, the SIP control block is allocated to the off-hook telephone set and the SIP control block is prevented from connecting to other telephone sets to establish a bi-directional digital voice packet transmission between the off-hook telephone set and the remote SIP number. When one telephone is using, the local SIP number for other telephones can establish new SIP control block with SIP call-leg to receive phone call and will not suffer to the blocked problem. 
   To achieve the above object, the present invention provides a method providing simultaneous call dialing and call receiving for multiple IP phones. The method combines the procedures mentioned in above two paragraphs and comprises following steps: A remote SIP number is dialed with at least one telephone set. At least one SIP control block containing an SIP call-leg is created for the local SIP number. The SIP control block is allocated to the dialing telephone set according to the SIP call-leg and to establish a bi-directional digital voice packet transmission between the dialing telephone set and the remote SIP number. When a remote SIP number calls the local SIP number, at least one SIP control block containing an SIP call-leg is created for a telephone group for making a ringing for all telephone sets in the telephone group. When a telephone set of the telephone group is off hook, the SIP control block is allocated to the off-hook telephone set to prevent the SIP control block from connecting to other telephone sets to establish a bi-directional digital voice packet transmission between the off-hook telephone set and the remote SIP number. The procedure is similar for a telephone group firstly receiving a phone call and then dialing a phone call. 
   The above summaries are intended to illustrate exemplary embodiments of the invention, which will be best understood in conjunction with the detailed description to follow, and are not intended to limit the scope of the appended claims. 

   
     BRIEF DESCRIPTION OF DRAWING 
     The features of the invention believed to be novel are set forth with particularity in the appended claims. The invention itself however may be best understood by reference to the following detailed description of the invention, which describes certain exemplary embodiments of the invention, taken in conjunction with the accompanying drawings in which: 
       FIG. 1  shows an application of the IP telephony apparatus according to the present invention. 
       FIG. 2  shows a block diagram of the IP telephony apparatus according to the present invention. 
       FIG. 3  shows an application of the present invention. 
       FIG. 4  shows a flowchart of dialing phone call through the IP telephony apparatus according to the present invention. 
       FIG. 5  shows a flowchart of receiving phone call through the IP telephony apparatus according to the present invention. 
       FIG. 6  shows the flowchart of communication procedure for the present invention. 
   

   DETAILED DESCRIPTION OF THE INVENTION 
     FIG. 1  shows an application of the IP telephony apparatus according to the present invention. The IP telephony apparatus according to the present invention provides busy-free communication for multiple VoIP telephone sets by using only one VoIP telephone number. The IP telephony apparatus according to the present invention adopts SIP VoIP protocol and can be an IP phone gateway, an ADSL modem, an ADSL router, a cable modem or a wireless network access point, etc. 
   The IP telephony apparatus according to the present invention supports two interface standards, namely, an FXS (Foreign eXchange Station) interface standard and an FXO (Foreign eXchange Office) interface standard. The FXS interface standard produces an FXS simulation signal for simulating a PSTN telephone line signal, and the FXS simulation signal can be connected to any PSTN telephony facility such as telephone set, wireless telephone set, fax or PBX. 
   The FXO interface standard generates an FXO simulation signal for simulating a PSTN facility signal and the FXO simulation signal can be connected to PSTN telephone line or extension line of PBX. In  FIG. 1 , the IP telephony apparatus according to the present invention has above-mentioned two interface standards and is connected between site A and site B. 
   In  FIG. 1 , the site A is a company in China and uses an FXO VoIP gateway  11  for accessing Internet  30  through an ADSL modem  12 . The FXO VoIP gateway  11  is connected to a PBX  13  through extension lines  14 ,  15  and the PBX  13  is connected to a plurality of PSTN telephone lines L 1 -L 5  and a pl urality of extension telephones  16 ,  17 . 
   In  FIG. 1 , the site B is a company in Taiwan and uses an FXS VoIP gateway  21  for accessing Internet  30  through an ADSL modem  22 . The FXS VoIP gateway  21  is connected to a plurality of telephone sets  24 ,  25  and connected to a PBX  23  through telephone lines L 6 -L 7 . The FXS VoIP gateway  21  is connected to a plurality of extension telephones  26 ,  27  and a plurality of PSTN telephone lines L 8 -L 12  through the PBX  23 . 
   When a user at site B makes a phone call to site A, the phone is transferred from Taiwan to China through Internet instead of international telephony network. The user at site B also can make phone call to other places in China through the PBX  13  at site A. In this situation, the user at site B only pays local-call fee, which is similar to domestic call fee in China. 
   When a user at site A makes a phone call to site B, the user at site A will hear dial tone and the telephone sets  24 ,  25  at site B will ring simultaneously. The user at site A (China) can make phone call to any place in Taiwan through the PBX  23  at side B and vice versa. This means that the a call in Taiwan can be transferred to site A through telephone line to PBX  23  and Internet, just like domestic call in Taiwan. 
     FIG. 2  shows a block diagram of the IP telephony apparatus  10  according to the present invention. As shown in this figure, the IP telephony apparatus  10  comprises a network connection port  41 , a plurality of telephone set connection ports  42 , an FXS interface unit  43 , a plurality of telephone connection ports  44 , an FXO interface unit  45 , a codec unit  46  and a VoIP allocation unit  47 . 
   The network connection port  41  can be an RJ-45 jack for connecting a network such as Internet or connecting a network through a network device. The telephone set connection port  42  can be a PSTN RJ-11 jack for connecting to telephony facilities such as telephone set, wireless handset, fax or PBX, etc. 
   The FXS interface unit  43  is connected between the telephone set connection ports  42  and the codec unit  46  and is used to produce an FXS simulation signal for simulating a PSTN telephone line signal. Therefore, the telephone set connection ports  42  can be directly connected to telephone set, fax or PBX telephone line. 
   The telephone connection ports  44  are connected to PBX extension line or telephone line of central exchange. The FXO interface unit  45  is connected between the telephone connection ports  44  and the codec unit  46  and is used to produce an FXO simulation signal for simulating a PSTN facility signal. 
   The codec unit  46  is a DSP unit to convert voice signal from telephone line or telephone into digital SIP packet and to convert digital SIP packet into voice signal. 
   The VoIP allocation unit  47  is an SIP processor and connected among the network connection port  41 , the FXS interface unit  43  and the FXO interface unit  45 . The VoIP allocation unit- 47  is functioned to process SIP packet for SIP VoIP phone call and allocate the connection between SIP service and physical phone. The VoIP allocation unit  47  establishes at least one control block in a local SIP number and the control block has a call leg. The call leg includes a call from field, a call to field and a call-ID field. 
   The VoIP allocation unit  47  allocates an SIP control block to any unused telephone set according to the call-leg and the telephone set allocated with the SIP control block can establish duplex voice packet transmission with external SIP number. 
     FIG. 3  shows an application of the present invention.  FIG. 4  shows a flowchart of dialing phone call through the IP telephony apparatus according to the present invention.  FIG. 4  shows a flowchart of receiving phone call through the IP telephony apparatus according to the present invention. 
   In this example, the IP telephony apparatus according to the present invention is used in site ( 1 ), site ( 2 ) and site ( 3 ) for accessing Internet. There are two phones (phone A and phone B) at site ( 1 ); three phones (phone O, phone P and phone Q) at site ( 2 ); and three phones (phone X, phone Y and phone Z) at site ( 3 ). In this example, each of the site ( 1 ), the site ( 2 ), and the site ( 3 ) is assigned with one SIP number. 
   As shown in  FIG. 3 , when the phone A at site ( 1 ) is off hook for dialing an SIP number at site ( 2 ) in step S 100 , the IP telephony apparatus  10  at site ( 1 ) finds an SIP number at site ( 1 ) by a mapped policy means for establishing communication link with the SIP number at site ( 2 ) in step S 102 . In this situation, there are multiple SIP numbers at site ( 1 ), and one of the multiple SIP numbers at site ( 1 ) is selected for communication. 
   The IP telephony apparatus  10  at site ( 1 ) then examines whether an unused control block is present in the selected SIP number in step S 104 . If an unused control block is not present, a new SIP control block is created in the SIP number at site ( 1 ) in step S 106  and the SIP control block has an SIP call-leg. The SIP call-leg includes a call from field, a call to field and a call-ID field. 
   If an unused control block is present, the step of creating new SIP control block is skipped and the unused control block is used to set new SIP call-leg. The SIP call-leg includes a call from field, a call to field and a call-ID field. 
   The SIP control block is allocated to the phone A at site ( 1 ) in step S 108  and SIP packet is sent to the IP telephony apparatus  10  at site ( 2 ) according to the SIP control block. 
   As shown in  FIG. 5 , the IP telephony apparatus  10  at site ( 2 ) receives the SIP packet and resolves the SIP call-leg therein. The IP telephony apparatus  10  at site ( 2 ) finds the SIP call-leg indicating the SIP number at site ( 1 ) calling an SIP number at site ( 2 ) in step S 200 . The IP telephony apparatus  10  at site ( 2 ) will examine whether an unused SIP control block is present in step S 202 . If an unused control block is not present, a new SIP control block is created in the SIP number at site ( 2 ) in step S 204  and the SIP control block has an SIP call-leg. 
   If an existing SIP control block is present or an unused SIP control block is present, the step of creating new SIP control block is skipped and the existing SIP control block is used to set new SIP call-leg. The SIP call-leg includes a call from field, a call to field and a call-ID field. 
   The IP telephony apparatus  10  at site ( 2 ) then allocates the SIP control block to the phone O, phone Pa and phone Q such that all the three phones will ring in step S 206 . At this time a ring-back signal is sent to the IP telephony apparatus  10  at site ( 1 ) and the user of phone A receives a ring-back tone. 
   When the phone O enters an off-hook state, the IP telephony apparatus  10  at site ( 2 ) ceases the ring tone for phone P and phone Q and allocates the SIP control block to phone O. The IP telephony apparatus  10  at site ( 2 ) further breaks the connection of the control block to phone P and phone Q and sends an Okay signal to IP telephony apparatus  10  at site ( 1 ) for permitting conversation. The IP telephony apparatus  10  at site ( 1 ) will respond an ACK (acknowledgement) signal to the IP telephony apparatus  10  at site ( 2 ). Afterward, the terminal at site ( 1 ) has bi-directional voice packet transmission with the terminal at site ( 2 ). Namely, the phone A at site ( 1 ) has telephony communication with the phone O at site ( 2 ).  FIG. 6  shows the flowchart of communication procedure between site ( 1 ) and site ( 2 ). 
   When the phone A at site ( 1 ) has telephony communication with the phone O at site ( 2 ), the phone B at site ( 1 ) still can dial an SIP number or receive phone call from another SIP number. The telephony communication of the phone B at site ( 1 ) will not be blocked by the telephony communication between the phone A at site ( 1 ) and the phone O at site ( 2 ). The phone B can establish telephony communication by creating new SIP control block with new SIP call-leg. 
   For example, when the phone B at site ( 1 ) dials an SIP number at site ( 3 ), all the phone X, phone Y and phone Z at site ( 3 ) will ring simultaneously. If the phone X at site ( 3 ) is off-hook, the telephony communication between the phone B at site ( 1 ) and the phone X at site ( 3 ) is established. At this time, the telephony communication between the phone A at site ( 1 ) and the phone O at site ( 2 ) is not bothered. 
   Similarly, when the phone Y at site ( 3 ) dials an SIP number at site ( 2 ), the IP telephony apparatus  10  at site ( 2 ) will create new SIP control block for the phone P and phone Q, while the phone O at site ( 2 ) is busy talking with phone A. The phone P and phone Q will ring and one of them establishes telephony communication with the phone Y at site ( 3 ). 
   Furthermore, when the phone Z at site ( 3 ) dials an SIP number at site ( 1 ), the users of phone A and phone B hear call-waiting tone and the users can switch their telephony communication to phone Z at site ( 3 ) by pressing a specific key such as transfer key or flash key. The specific key is pressed again to return to original communication. More over, the users phone A and phone B can press special key to cease the connection of the SIP control block from the phone Z, Therefore the users of phone A and phone B will not be bothered by the new coming call. 
   Although the present invention has been described with reference to the preferred embodiment thereof, it will be understood that the invention is not limited to the details thereof. Various substitutions and modifications have suggested in the foregoing description, and other will occur to those of ordinary skill in the art. Therefore, all such substitutions and modifications are intended to be embraced within the scope of the invention as defined in the appended claims.