Patent Application
20080123627
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
