Abstract:
Systems and methods are disclosed for a media terminal adapter (MTA) that includes a session initiation protocol (SIP) to media gateway control protocol (MGCP) translator. The MTA receives SIP-based signaling packets including the MTA address and subsequently translates the signal packets to provide MGCP-based signaling packets. The MGCP-based signaling packets are subsequently transmitted to a communications network in order to set up a call where the associated voice packets are transmitted with QoS.

Description:
FIELD OF THE INVENTION 
       [0001]    This invention relates in general to telephony systems over broadband coaxial cable, and more particularly, to the field of enabling a session initiation protocol proxy in a media terminal adapter. 
       DESCRIPTION OF THE RELATED ART 
       [0002]    Media terminal adapters (MTAs) are the interface to the physical telephony or video equipment required for voice over Internet Protocol (VoIP) transport. Today, Data over Cable Service Interface Specification (DOCSIS) VoIP gateways, or embedded MTAs (EMTAs), which include both an MTA and a cable modem, provide quality of service (QoS) to voice calls that are generated by phones connected directly to the MTA. QoS is used to create quality of service transport guarantees for voice packets dynamically on a per call basis. QoS is used in the networks to ensure low latency and guaranteed bandwidth for voice packets typically using Real Time Protocol (RTP) for each phone call on the DOCSIS network. Since the DOCSIS network can become congested, QoS is used to ensure that VoIP calls are not impacted. When not needed for phone calls, the bandwidth that is not needed by high priority QoS packet flows can be used for lower priority packet flows such as web surfing and e-mail. MTAs using media gateway control protocol (MGCP) make use of significant infrastructure investment in MGCP equipment including support for QoS, MGCP softswitches, and provisioning servers. This infrastructure exists to ensure that MGCP-based phone calls receive preferred quality of service on the DOCSIS network and to control the packet switching of phone calls to MTA phone line endpoints and assign one or more phone numbers to each MTA endpoint. 
         [0003]    Users may now use a session initiation protocol (SIP) phone, such as a WiFi (wireless fidelity) phone or a personal computer (PC) based phone. When the SIP-based phones are used with a conventional EMTA or cable modem for VoIP service, the audio phone call is carried over the DOCSIS network without the benefit of using any of the MGCP infrastructure available for MGCP phone calls. More specifically, the SIP-based phone calls face several limitations or restrictions. Users now making a call to or from a SIP-based phone are not able to use QoS so the voice packets from SIP-based phone calls compete with other Internet traffic, such as e-mail or web browsing, for bandwidth. Therefore, there is a need for a system and method that allows a SIP-based phone connection over the DOCSIS network while maintaining a QoS that is expected by the users. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0004]    The invention can be better understood with reference to the following drawings. The components in the drawings are not necessarily drawn to scale, emphasis instead being placed upon clearly illustrating the principles of the invention. In the drawings, like reference numerals designate corresponding parts throughout the several views. 
           [0005]      FIG. 1  illustrates a communications system including a conventional telephone and a PC connected to an MTA for transporting voice and data packets over a communications network. 
           [0006]      FIG. 2  illustrates a communications system including the conventional telephone, a SIP-based PC phone, and a WiFi SIP phone connected to an MTA for transporting packets over the communications network. 
           [0007]      FIG. 3  illustrates a communications system including the conventional telephone, a SIP-based PC phone, and a WiFi SIP phone connected to an MTA including an SIP to MGCP translator in accordance with the present invention. 
           [0008]      FIG. 4  illustrates a processor within the MTA with the SIP to MGCP translator in accordance with the present invention. 
           [0009]      FIG. 5  illustrates routing information attached to generated SIP-based signaling packets based on conventional routing information and present invention routing information. 
       
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
       [0010]    Preferred embodiments of the invention can be understood in the context of a broadband communications system. Note, however, that the invention may be embodied in many different forms and should not be construed as limited to the embodiments set forth herein. All examples given herein, therefore, are intended to be non-limiting and are provided in order to help clarify the description of the invention. 
         [0011]    The present invention is directed towards a system and method for transmitting voice packets having QoS that are generated from SIP-based telephones over a DOCSIS communications network. Importantly, the SIP-based phone calls can use network infrastructure designed for MGCP-based phone calls. More specifically, an MTA receives SIP call signaling packets and subsequently translates the SIP call signaling packets into MGCP call signaling packets. The translated MGCP call signaling packets then set up QoS with security for the voice RTP packets. This is advantageous over the conventional method of routing voice packets from SIP-based telephones where the SIP voice packets compete for bandwidth with other Internet traffic and are unable to use the infrastructure that is available to MGCP voice packets. MGCP voice packets that are received from a conventional telephone are also transmitted through the MTA having QoS in a known manner. 
         [0012]      FIG. 1  illustrates a communications system  100  including a conventional telephone  105  and a PC  110  connected to an MTA  115  for transporting voice and data packets over a communications network  120 . The telephone  105  is physically connected to the MTA  115  using standard wiring and telephone jacks, such as CAT-3 and RJ11 connectors. Voice signals received from the telephone  105  are packetized by the MTA  115 . The voice packets are then transmitted over the communications network  120  using an MGCP protocol over DOCSIS to a cable modem termination system (CMTS). Importantly, the voice packets are transmitted over the communications network  120  having QoS, which is illustrated by the dotted lines between the MTA  115  and the communications network  120 . 
         [0013]    The PC  110  is generally connected to the MTA  115  with an Ethernet cable and Ethernet plugs and jacks although it may also be connected with a wireless gateway. Data packets are transmitted to and received from the MTA  115 . The data packets are transmitted and received from the communications network  120  using Internet addresses in a known manner. The data packets, such as e-mail and web browsing, are transmitted over the communications network  120  with a best effort. In other words, the Internet traffic, which is enabled by an Internet Services Provider (ISP), does not have QoS, which is illustrated by the solid lines between the MTA  115  and the communications network  120 . 
         [0014]      FIG. 2  illustrates a communications system  200  including the conventional telephone  105 , a SIP-based PC phone  205 , and a WiFi SIP phone  210  connected to the MTA  115  for transporting signaling, voice, and data packets over the communications network  120 . The SIP-based signaling packets set up the call; for example, dialing a telephone number and setting up the call by using a session description protocol (SDP), which describes where the voice packets are being transmitted. The voice packets are then transmitted via RTP packets the intended receiver. In this implementation, the signaling, voice, and data packets include a destination Internet address of the intended receiving telephone or computer, and are transmitted over an Ethernet cable to the MTA  115 . The MTA  115  then forwards the packets to the communications network  120 , which are then combined with all the Internet traffic with only a best effort. 
         [0015]    The WiFi SIP phone  210  generates signaling and voice packets, including a destination address of an intended receiving telephone or computer, and are transmitted and received by an antenna (not shown) in the MTA  115 . The MTA  115  then forwards the signaling and voice packets to the communications network  120 . In this manner, the SIP signaling sets up the call, and the voice packets are then combined with other Internet traffic with only a best effort. Disadvantageously, the voice packets without QoS may be dropped at any time or delayed during the telephone conversation, which degrades the quality of the voice communication heard by both the caller and the receiver. 
         [0016]      FIG. 3  illustrates a communications system  300  including the conventional telephone  105 , the SIP-based PC phone  205 , and a WiFi SIP phone  210  connected to an MTA  315 , where the MTA  315  includes an SIP to MGCP translator in accordance with the present invention. The telephone  105  transmits MGCP-based signaling packets, and the voice packets are transmitted in the same manner as described above in connection with  FIGS. 1 and 2 . The SIP-based PC phone  205  generates SIP-based signaling packets that are transmitted to the MTA  315 . In accordance with the present invention, the MTA  315  translates the SIP-based signaling packets to MGCP-based signaling packets. The translated MGCP-based signaling packets then set up the call for the voice RTP packets using the MGCP infrastructure including security parameters. The voice RTP packets are subsequently transmitted over DOCSIS with QoS, which is illustrated by the dotted lines connecting the MTA  315  and the communications network  120 . SIP-based data signals generated by the PC  205  are routed over the communications network  120  via the MTA  315  with a best effort and are represented as the solid lines. Additionally, SIP-based signaling packets generated by the WiFi SIP phone  210  are transmitted to the MTA  315 , which then translates the SIP-based signaling packets to MGCP-based signaling packets. The translated MGCP-based signaling packets then set up the call having QoS for the voice RTP packets, which follows the dotted line connecting the MTA  315  and the communications network  120 . 
         [0017]      FIG. 4  illustrates a processor  400  within the MTA  315  with the SIP to MGCP translator in accordance with the present invention. The processor  400  includes software and hardware for translating the SIP-based signaling packets into the MGCP-based signaling packets. A first receiver point  405  is coupled to the conventional telephone  105  that is used for generating MGCP-based signaling and voice packets. A second receiver point  410  is coupled to the SIP-based PC phone  205  that receives SIP-based signaling, voice, and data packets. A third receiver point  415  is coupled to the WiFi SIP phone  210  that wirelessly receives SIP-based signaling and voice packets. The SIP-based signaling packets received from both the SIP-based PC phone  205  and the WiFi SIP phone  210  are translated to MGCP-based signaling packets by the SIP to MGCP translator. After translation, the MGCP-based signaling packets then set up the call using the MGCP infrastructure including QoS. The voice packets are associated with the translated MGCP-based signaling packets are routed to the communications network  120  having QoS, which is illustrated by dotted line  420 . As mentioned, the data packets continue transmission through the communications network  120  with a best effort, which is illustrated by the solid line  425 . 
         [0018]      FIG. 5  illustrates routing information attached to generated SIP-based signaling packets based on conventional routing information and present invention routing information. Conventionally, the SIP-based signaling packets  505  include a destination Internet address in attached header information. Accordingly, the SIP-based signaling packets  505  for setting up the call are forwarded via the MTA  115  ( FIG. 2 ) to the intended receiver using the destination Internet address. In this manner, the SIP-based signaling packets  505 , and subsequently, the voice packets are routed with only a best effort (i.e., without QoS). 
         [0019]    In accordance with the present invention, however, the destination address for generated SIP-based signaling packets  515  now reflects an address associated with the MTA  315 . The destination address of the MTA  315  is programmed into the PC  205  and the WiFi phone  210  either by a user of the equipment or a service provider. When the MTA  315  receives the SIP-based signaling packets  515  including its address as the destination, the MTA  315  provides the SIP-based signaling packets  515  to the SIP to MGCP translator  400  for conversion. Subsequently, the translated MGCP-based signaling packets then set up the call using the MGCP infrastructure for the voice packets. The SIP-based data packets  525  from the PC  205  include an Internet destination address  530  so that the MTA  315  continues to forward these packets  525  to the communications network  120  with a best effort. 
         [0020]    Accordingly, systems and methods have been provided that allows transmission of SIP-based voice packets having QoS. It will be appreciated that further embodiments are envisioned that implement the invention, for example, using all software or adding modes for additional features and services.