A. Technical Field
The present invention relates generally to computer network protocols, and more particularly, to addressing and delivery of voice packets within a Voice over Internet Protocol (VoIP) connection.
B. Background of the Invention
The popularity of VoIP as a method for providing telephone service across networks is continually increasing. VoIP systems provide telephone connections by transmitting voice packets between two telephone devices via a packet-switched network (e.g., TCP/IP network). This increase in VoIP popularity is primarily due to two reasons: the relatively inexpensive cost of a VoIP telephone call and recent networking advancements causing an increase in the quality of VoIP communication.
VoIP lets service providers offer long-distance services to clients at much lower rates than traditional phone companies. VoIP also uses networks more efficiently than the traditional public switched telephone network used by the traditional phone companies. One reason for this increase in efficiency is the ability of VoIP to time-division multiplex voice data (i.e., telephone connections) together on a single line within a network. Thus, the bandwidth utilization increases within a packet switched network allowing more telephone connections to occur simultaneously.
A few years ago, the quality of a VoIP connection was lacking due primarily to packet delay occurring as voice packets traveled across these networks. This problem was primarily caused by the inefficiency of the Internet over which the VoIP connections occurred. Internet events such as bottlenecks, jitters and discarding packets reduced the quality of a VoIP telephone conversation occurring across the Internet. However, the increase of large private networks, more controlled Internet backbones, and more efficient routing protocols have greatly reduced these problems. Accordingly, the quality of a VoIP telephone conversation today has drastically improved. Some providers have also chosen to avoid the public Internet because of the difficulty in ensuring end-to-end control of service quality. These providers have created managed networks on which VoIP connections may be easily controlled and new VoIP technology may be more easily implemented. As the popularity of VoIP continues to grow, other issues need to be addressed, such as security, network interoperability and compatibility, to ensure the future success of VoIP.
The telephone connection is established by the first gateway 110 receiving a connection request from the first telephone 105 that includes a destination telephone number. This destination telephone number may be a ten-digit telephone number similar to those used over traditional publicly switched telephone networks. In response, the first gateway 110 requests a destination network address from the gatekeeper 140 corresponding to the destination telephone number. This conversion allows the first gateway 110 to locate the second gateway 120 on the Internet 130. Typically, this conversion results in a network address, such as an IP address that differentiates the second gateway 120 from other gateways on the Internet 130.
A set-up procedure is initiated by the first gateway 110 in which the second gateway 120 is provided the address of the first gateway 110. This set-up procedure results in a connection on which data, particularly voice packets and control data, are transmitted between the gateways 110, 120. This data may travel through multiple networks and multiple routers/switches within these networks in order to reach the correct destination. As described above, oftentimes the quality of this connection is lacking due to the characteristics of the Internet 130. Congestion and failures, within these networks, may drastically reduce the rate at which this data travels in an established connection and may increase the number of packets that are lost or discarded prior to reaching a particular destination address.
The established connection between the first gateway 110 and the second gateway 120 presents various security concerns. A large number of these issues are caused by the visibility of the gateways 110, 120 within the connection. Specifically, the IP addresses of the gateways 110, 120 are known by each other. This visibility compromises the security of all of the devices attached to a network having a visible gateway. Accordingly, a hacker may access devices on the network, other than the telephone or computer participating in the connection, through the gateways 110, 120. For example, after gaining access to the network through a gateway 110, 120, a hacker may access an unauthorized networked device through techniques such as IP spoofing or other commonly used hacking methods. Accordingly, network providers prefer to mask their gateway addresses from outside devices in order to further secure the network against hacking and other unauthorized access to their networks.
The first telephone 105 is connected to a first network gateway 212(a) via first analog connection 107. This first network gateway 212(a) resides in a large private network 210 that contains multiple gateways 212(a)-(d). The second telephone 115 is connected to a second network gateway 222(a) via second analog connection 117. This second network gateway 222(a) resides in a second large private network 220 that also contains multiple gateways 222(a)-(d). The first gateway 212(a) is coupled to a first proxy 235 and the second gateway 222(a) is coupled to a second proxy 240.
The first and second proxies 235, 240 hide the addresses of the first and second gateways 212(a), 222(a) from each other. Specifically, the first proxy 235 is aware of the network addresses of the first gateway 212(a) and the second proxy 240, but not the second gateway 222(a). The second proxy 240 is aware of the network addresses of the second gateway 222(a) and the first proxy 235, but not the first gateway 212(a). Thus, communication between devices on the first network 210 and the second network 220 occur through the proxies 235, 240 while maintaining a level of privacy from each other.
The first and second proxies 235, 240 require that packets traveling through the VoIP connection may be modified multiple times. Specifically, in order for the first and second proxies 235, 240 to extract and analyze information from a packet header (e.g., port number). Once this information is extracted, a new header is usually put on the packet and it is compressed. Thereafter, the packet is transmitted from a proxy. Because voice packets travel through multiple proxies 235, 240, the number of packet manipulation operations increases. Thus, there is a need to reduce the number of proxy devices within a VoIP connection. This need is further highlighted by the high cost of networking devices such as proxy devices.
Communication between the first proxy 235 and the second proxy 240 may occur using an IP suite protocol implementing either TCP or UDP depending on the type of data within packets. UDP is generally used for VoIP telephone connections due to the time sensitivity of the VoIP connection. Accordingly, sockets are established between the first and second proxies 235, 240. A socket is a combination of an IP address and a port that creates a device-to-device path on which packets may be transmitted and received. Thus, a proxy or other networking device may have numerous ports that provide communication paths on which packets may travel.
Oftentimes, a simple packet translation method will not properly switch a voice packet along a VoIP connection. For example, this switching process may be complicated if the networks on which the first and second gateways 212(a), 222(a) are not directly compatible. Generally, voice traffic is transmitted according to the H.323 standard, an ITU real-time standard for transmission of voice over networks. However, there are variations in the implementation of the H.323 standard by network providers that may cause incompatibilities between networks. These variations often require packet modification operations to occur within a proxy to provide smooth voice traffic between the incompatible networks.
In order to perform packet translation and switching operations in connections between to directly incompatible networks, a proxy must be able to identify the type of network from which the packet was sent and to which the packet is destined. Also, the proxy must be able to identify the packet type (e.g., RTP) in order to perform packet translation and switching operations. Once this information is identified, the proxy may modify the packet so that it is able to effectively travel through a network to a destination gateway.
As previously described above, it is important to try and reduce the number of switches, routers and other networking devices within a VoIP connection for two primary reasons. First, networking devices are expensive and the initial cost as well as the management cost may be significant. Second, each networking device increases the possibility of errors such as packets being discarded or failure as well as causes an additional delay within a VoIP connection. As a result, researchers have been developing technology that reduces the number of networking devices within a network.
Accordingly it is desirable to provide network address translation within a network device that masks both ends of a VoIP connection from each other. Additionally, it is desirable to provide network address translation within a network device that facilitates VoIP connections between different types of networks and that processes different types of packets within a VoIP connection. Furthermore, it is desirable to provide network address translation within a network device that increases the number of VoIP connections that may be served by the network device. It is additionally desirable to allow the dynamic allocation of ranges of ports within the network device for use with a VoIP connection. It is still further desirable to allow components of the network device to learn gateway addresses in the presence of intervening firewalls.
The present invention overcomes the deficiencies and limitations of the prior art by providing a set of commands used by a voice connector for dynamically allocating port ranges within a voice switch. The present invention further accounts for network address translation performed by intermediary network devices such as firewalls by providing a Connect command used by the voice connector to inform the voice switch of possible addresses used by the intermediary network device. The voice switch then applies bitmasks specified by the Connect command to incoming voice packets from unknown addresses to determine whether the packets are being sent from known firewalls, in which case the packets are accepted and corresponding addresses learned.
In one embodiment, a voice router for dynamically allocating port ranges comprises a voice connector adapted to initiate a connection between a first network gateway and a second network gateway and a voice switch adapted to route voice data between the first network gateway and the second network gateway. The voice connector sends a request for a plurality of ports to the voice switch and, responsive to the request, the voice switch allocates a number of ports, creates a corresponding entry in a port range table, and provides to the voice connector a response comprising an identifier of the port range.
In another embodiment, a voice router for routing packets in the presence of a firewall comprises a voice connector adapted to initiate a voice connection between a first network gateway and a second network gateway and a voice switch adapted to route voice data between the first network gateway and the second network gateway. The voice connector sends a connect packet to the voice switch, the connect packet specifying the IP address of the first network gateway and an address bitmask. Responsive to the voice switch receiving from the first network gateway, via the firewall, a first voice packet having a first packet source IP address, the voice switch computes a first masked value resulting from a bitwise AND of the address bitmask and the first packet source IP address, and computes a second masked value resulting from a bitwise AND of the address bitmask and an IP address of the first network gateway. Responsive to the first masked value being equal to the second masked value, the voice switch stores the first packet source IP address in association with the IP address of the first network gateway and forwards the first voice packet to the second gateway.
The features and advantages described in this summary and the following detailed description are not all-inclusive, and particularly, many additional features and advantages will be apparent to one of ordinary skill in the art in view of the drawings, specification, and claims hereof.
The figures depict a preferred embodiment of the present invention for purposes of illustration only. One skilled in the art will recognize from the following discussion that alternative embodiments of the structures and methods illustrated herein may be employed without departing from the principles of the invention described herein.
The present invention describes a network router/bridging device that interfaces networks within a VoIP connection and masks the location of each network from the other. This device is able to interface networks implementing different H.323 protocols (or SIP protocol) and reduces the number of ports required for this connection. Specifically, the device provides a novel network address translation module and network-type identification module that facilitates a VoIP connection between these networks. The novel network address translation module increases the number of VoIP connections the networking device may route or switch by reducing the number of ports required for each individual connection. According to one embodiment of the invention, only one port on the networking device is required for each VoIP connection. This reduction in the number of required ports is provided by an address translation that is able to set-up a VoIP connection on a single port that determines the direction of a packet received at a single bi-directional port and able to identify the type of network to which the packet is destined.
A. VoIP Connection Using a Single Voice Router
A first port configuration according to the present invention between the two proxies 335, 340 is shown. Audio communication between the two proxies 335, 340 occurs over 4 ports. A first port, port N 310, receives Real-time Transport Protocol (RTP) packets at the first proxy 335 from the second proxy 340. A second port, port M 315, receives RTP packets at the second proxy 340 from the first proxy 335. Ports N 310 and M 315 may also be assigned the same port number or different port numbers depending on the implementation of the connection. A third port, port R 330, receives Real-time Transport Control Protocol (RTCP) packets at the first proxy 335 from the second proxy 340. A fourth port, port S 325, receives RTCP packets at the second proxy 340 from the first proxy 335. Port R 330 and port S 325 may be assigned the same port number or different port numbers depending on the implementation of the connection. Each of these ports (i.e., N, M, R, and S) along with an IP address of a corresponding proxy (i.e., 335 or 340) creates a socket on which packets flow. Thus, the proxies 335, 340 may identify the source of a packet by listening on the specific port number corresponding to the transmitting source. The proxies 335, 340 have a limited number of ports and addresses that they may use. Accordingly, as the number of ports that are used for each connection increases, the number of total connections that a proxy can serve decreases.
After one of the proxies 335, 340 receives a packet, it will forward the packet onto a corresponding gateway 312, 322 via an additional port. For example, a RTP voice packet received by the first proxy 335 on port N 310 is forwarded on to the first gateway 312 on port A 350. Comparatively, an RTP voice packet received by the second proxy 340 on port M 315 is forwarded on to the second gateway 322 on port B 360. A similar method is used for RTCP packets where packets transmitted onto port R 330 for the first proxy 335 and on port S 325 for the second proxy are forwarded onto a corresponding gateway via particular ports (not shown). The usage of ports by proxies and gateways can vary depending on the design of the private networks and network interconnectivity.
The voice router 400 also manages RTCP packet streams between the first and second network gateways 212(a), 222(a) in a similar manner. Specifically, RTCP packets from the first network gateway 212(a) arrive on port R 425 at the voice router 400. These packets are forwarded onto the corresponding port (port S 420) of the voice router 400 and then to the second network gateway 222(a). RTCP packets from the second network gateway 222(a) are received on port S 420 at the voice router 400. These packets are then forwarded onto the corresponding port (port R 425) of the voice router 400 and then to the first network gateway 212(a).
This embodiment of the present invention does not require the module responsible for the H.323 protocol (or SIP protocol), in particular the H.245 logical channel negotiation, to inform the voice router 400 of the IP addresses and ports used for both the gateways 212(a), 222(a). As a result, the information exchange between the H.323 protocol handling module and voice router 400 is reduced.
This four-port configuration allows the voice router 400 to identify the direction of the packet streams between the first network gateway 212(a) and the second network gateway 222(a) by the port on which a packet arrives. Packet direction is discovered by identifying the source of the packet and using the source identification to determine a destination corresponding to the source. Specifically, the voice router 400 is aware of the connections that it serves and may determine a destination of a packet by identifying the port on which the packet arrived. Additionally, the four-port configuration provides for communication between the two network gateways 212(a), 222(a) in two different protocols, namely RTP and RTCP. However, as described above, this high port count also limits the number of VoIP connections that the voice router 400 can support.
According to one embodiment, the ports may be assigned according to the types of networks involved in the VoIP connection. For example, if the voice router 400 is positioned between a private network and the public Internet, then the port assignment may occur in the following manner. The first port is pre-assigned for the RTP stream originating from a gateway on the public Internet. The second port is pre-assigned for the RTCP stream originating from the gateway on the public Internet. The third port is pre-assigned for the RTP stream originating from a gateway in the private network. The fourth port is pre-assigned for the RTCP stream originating from the gateway in the private network.
Referring to the above described port configuration, when the voice router 400 receives a first UDP packet of the RTP stream from the private network, it reads the source IP address and port number within the UDP packet. This address is the transmitting gateway's IP address and the port number is the port on which the packet arrived. In this example, the packet originated at a private network, and therefore, the port number would be the third port number within the tuple, as described above. This private gateway address is stored with the voice router 400 and will be used to transmit packets from the public Internet to the correct destination private network within the particular VoIP connection. A similar method may be used when an RTP packet arrives from the public Internet destined to a particular private network. As a result of this process, the voice router 400 is able to help create and maintain a VoIP connection.
a) Port Reduction on the Single Voice Router
The voice connector 570 is used to set-up a VoIP connection. The voice connector 570 may be integrated within the voice switch 585, as shown in
A call set-up request contains information regarding the desired VoIP connection including destination information such as an address. An address, such as a ten-digit telephone number, within this request is translated by the voice connector 570 into a destination IP address. This translation may occur by accessing a gatekeeper that is either public or operating within a private network and is masked from the requesting gateway. Once this address translation occurs, the voice connector 570 creates a virtual connection between the two network gateways 212(a), 222(a) by assigning a port or ports on which this VoIP communication will occur. Once these ports are assigned, this information is transmitted to the voice switch 585 and to the network gateways 212(a), 222(a). Sockets, an IP address and port number, are established between the different networking devices and the voice switch 585. Voice packets are then transmitted between the first and second gateways 212(a), 222(a) on these sockets.
The VoIP connection may also comprise networking devices that may adjust the connection configuration and port number assignments within the connection. For example, firewalls or other network servers having corresponding addresses and port numbers may be included to enhance security or add other functionality within the connection. Accordingly, the number of sockets may increase within the connection to facilitate the inclusion of these devices.
The voice switch 585 is able to identify these packets by extracting source information contained within the packet header. Specifically, the voice switch 585 extracts and analyzes the IP source address within the packet header in order to correctly switch the packet to the correct network gateway. This analysis may be done using a number of different methods. For example, the extracted IP source address may be compared to an IP address of either the first network gateway 212(a) or the second network gateway 222(a). If the extracted IP source address matches the compared network gateway address (address for gateway 212(a)), then the packet is forwarded accordingly (e.g., to gateway 222(a) with a new header having a source IP address of the voice router 500 and a destination IP address of gateway 222(a)). However, if the two addresses do not match, then packet is forwarded to the other network gateway (e.g., to gateway 212(a)) by default because only two possible destination gateways exist within the VoIP connection. Specifically, a buffer may be maintained within the voice switch 585 that maintains these two addresses to which a source address in a packet header is compared. Thus, the voice switch 585 is able to reduce the number of ports required to maintain a RTP VoIP connection and still maintain correct packet flow within this connection with the implementation of this novel address translation. This packet direction identification is discussed in greater detail below with reference to
The voice switch 585 also reduces the number of RTCP ports on the VoIP connection. Specifically, as with RTP port reduction, the number of required RTCP ports is reduced from two to one for each voice connection. The RTCP protocol is a companion protocol to RTP and is used to provide control and quality of service data to various devices within a connection. There are typically less RTCP packets transmitted by a gateway than RTP packets. The functionality provided by RTCP data may be compensated, at least to a particular level, internally within the voice switch 585. Because so few RTCP packets are transmitted from a gateway and the lost functionality of discarded RTCP packets may be minimized by the voice switch 585, discarded RTCP packets typically do not have a significant effect on the quality of a VoIP connection using the voice switch 585.
The voice switch 585 discards RTCP packets after they are received on port M 510 in order to further reduce the port count of a VoIP connection. Because the number of RTCP connections is reduced from two to one for each voice connection, the number of available ports on the voice switch 585 drastically increases. It is important to note that the voice switch 585 needs to have at least one RTCP port 510 on which RTCP packets arrive. For example, if the voice switch 585 did not have the RTCP port 510, then bounce-backs or acknowledgements would be transmitted from the voice switch 585 to a gateway transmitting RTCP packets. This bounce-back or acknowledgement would signal the transmitting gateway that there are no available RTCP ports on the voice switch 585. This acknowledgement presents a security risk to the voice switch 585 and attached network because hackers would be able to listen to particular ports on the voice switch 585. Thus, the single aggregating RTCP stops this acknowledgement and increases security on the voice switch 585.
The network type identification module 550 identifies a network type corresponding to a packet's destination gateway and source gateway. This identification also allows the voice switch 585 to ensure that the transmitted packet is compatible with the destination gateway (e.g., a packet is transmitted on the correct port number and to the correct destination port). There are multiple methods by which these gateways may be identified. First, this information may be embedded within header fields, such as port numbers, within the packet. Second, ports may be assigned according to the types of gateways in the VoIP connection. Both of these methods are described in greater detail below.
The packet translation module 580 receives information regarding the source and destination network types and the packet direction in order to correctly identify an appropriate packet translation operation(s). The packet translation module 580 ensures that the packet is transmitted on the correct port number so that it is compatible with the destination gateway. Specifically, the packet translation module 580 inserts the correct IP address of the destination gateway and the correct port number on the destination gateway within the packet header. The prior packet header may have been already discarded or be discarded by the packet translation module 580 prior to or after a new header is placed on the packet. These packet translation operations will be described in greater detail below.
A network address translation module 598 is implemented within the voice connector 570 to provide translation of a call set-up request in order to properly establish a VoIP connection. This translation may require accessing an external gatekeeper or may be done internally within the voice connector 570. As described above, the network address translation module 598 receives a request from a gateway 212 or 222 to make a connection. This request typically identifies the other side of the connection by a ten-digit telephone number or other identifying number. The network address translation module 598 uses a database to translate this ten-digit telephone number to an IP address. This translation may be done internally within the network address translation module 598 or may be done externally by addressing a public or private gatekeeper to translate the telephone number to an IP address. As a result of this process, the voice connector 570 will have identified the IP addresses of both the requesting gateway (i.e., from the call set-up request) and the destination gateway (i.e., from the above-described translation).
A port initialization module 595 within the voice connector 570 is used to assign ports to particular VoIP connections. The port initialization module 595 communicates with the network address translation module 598. In response to a VoIP connection request, the port initialization module 595 assigns a port or ports on which packets will travel between the two gateways. This port information is then transmitted to both gateways, for example, using port G 565 and port F 575, and the voice switch 585 via line 588. Accordingly, the voice switch 585 will be able to identify a packet by listening on a particular port(s). For example, the first and second gateways 212(a), 222(a) are told to transmit voice packets on port N 505 to the voice switch 585. This port information is also transmitted to the voice switch 585 along line 588. As a result, the voice switch 585 is able to identify packets within this particular connection by listening on port N 505.
The port initialization module 595 may assign these ports in relation to the gateway types of the gateways 212, 222. The port initialization module 595 may access a network pair table 590 in order to assign ports from port ranges corresponding to the gateway type connections. For example, if the first gateway 212(a) has a first gateway type (e.g., Cisco, Clarent, etc.) then the port initialization module 595 may select a port from a range of ports (e.g., 3000-4000) corresponding to that first gateway type. Thereafter, when voice packets are actually transmitted on these ports within the connection, the voice switch 585 can identify the gateway type of the gateway/network that transmitted the packet. In one embodiment, the voice connector 570 does not rely on a static network pair table 590 to provide the port ranges, but instead uses a port allocation module 599 to establish the port ranges on demand, as further described below.
In another embodiment, the port initialization module 595 may assign these ports in relation to the physical locations of the gateways. For example, the network pair table 590 may contain ranges of port values corresponding to physical locations of gateways. Thus, if the first gateway 212(a) is physically located in China, then the port initialization module 595 may select a port from a range of ports (e.g., 5000-6000) corresponding to that physical location. Thereafter, when voice packets are actually transmitted on the ports within the connection, the voice switch 585 can identify the physical location of the gateway/network that transmitted the packet.
As described above, VoIP connections between multiple networks typically follow the H.323 standard. However, various interpretations of this standard by network service providers may present compatibility issues between two separate networks. For example, a Clarent H.323-based network may have difficulty directly mapping to a Cisco H.323-based network due to slight protocol variations between the two networks. For example, port assignment protocols between a Cisco H.323-based network and the voice switch 585 may differ from those between a Clarent H.323-based network and the voice switch 585. In such an occurrence, the voice switch 585 may perform an additional step within the packet translation operation (e.g., compensate for differing port assignment protocols) between the two networks in order to ensure proper communication. In order to correctly perform this translation, the voice switch 585 should identify both the gateway type of the gateway from which the packet was sent and the gateway type of the gateway to which the packet is destined.
The first network gateway 212(a) resides on the first network 210 with corresponding first network type—e.g., a Clarent network. This first type of network requires that H.323 compatible packets be transmitted from a port X on the first gateway 212(a) to a port N 505 of the voice switch 585 and for H.323 compatible packets to be transmitted from the port N 505 on the voice switch 585 to the a port X+2 on the first gateway 212(a). In comparison, the second network gateway 222(a) resides on the second network 220 with corresponding second network type—e.g., a Cisco network. This second type of network requires that H.323 compatible packets be transmitted from a port Y of the second gateway 222(a) to port N 505 of the voice switch 585 and for H.323 compatible packets to be transmitted from port N of the voice switch 585 to the same port Y of the second gateway 222(a). Thus, although both the first and second networks follow the H.323 standard, differing interpretations of this standard have led to different port assignment procedures between the two networks. In order to compensate for this difference, the voice switch 585 of the present invention is able to identify these slight variations between networks when both assigning port numbers during the call set-up procedure and packet switching as the telephone call is occurring.
The voice switch 585 is able to compensate for these variations between networks and properly translate packets between the two networks within the connection (e.g., transmit a packet on the appropriate port number to a network). First, the voice switch 585 identifies the direction of a packet within a connection (i.e., identifies a destination address for the packet). Second, the voice switch 585 identifies the type of network to which a packet is destined. This information allows the voice switch 585 to ensure that a packet transmitted from the voice switch 585 is compatible with the network to which the packet is destined. Moreover, the present invention also reduces the number of ports used from 4 to 2 as compared with the prior art.
b) Network Type Identification
As described above, the voice router 500 needs to be aware of the types of networks in this VoIP connection in order to ensure that proper packet translation occurs. Once the types of the two networks are identified, an appropriate packet translation may be retrieved and performed accordingly.
(i) Network Type Internal Table
A first method that may be used to identify network types is embedding a network type identifier within a port number found in the packet header.
During the set-up of the VoIP connection, addressing information is gathered and the IP addresses and types of the first and second networks are determined. Thereafter, the two IP addresses are compared to identify the smaller IP address and the larger IP address. The comparison provides an order in which the network type identifiers corresponding to the two networks will be inserted within the port number 600. A first bit mask is inserted in the first two-bit field 610. The second bit mask is inserted in the second two-bit field 620. Thus, when the voice switch 585 extracts these two bit masks; it will be able to associate each identifier to a particular network through the position (e.g., field 610 or 620) from which the identifier was taken. For example, the first bit mask 610 contains network type information for the smaller IP addressed network and the second bit mask 620 contains network type information for the larger IP addressed network. Thereafter, a port value is assigned and inserted within the port value field 630. The port number 600 for the packet is the combination of these three fields. The port number 600 is inserted within the packet header and the packet is transmitted to the voice switch 585. Although this process results in reducing the number of available ports per voice switch 585 on which packets may be transmitted (i.e., due to the allocation of four of the 16 bits to bit masks), the resulting reduction in the number of ports per connection on the voice switch 585 more than compensates for this reduction.
The voice switch 585 extracts this port number 600 from the packet header after receiving the packet on a corresponding port. From this port number 600, the network type information within the first and second network type identifiers 610, 620 is removed and analyzed. From this information, the voice switch 585 is able to identify the network types of both networks within the VoIP connection. According to the example discussed above, the voice switch 585 extracts the network type information from the first network type identifier 610 and assigns this network type to the network with the smaller IP address. The information within the second network type identifier 620 is extracted and assigned to the network with the larger IP address. Thereafter, a corresponding packet translation operation is performed on the packet prior to transmission to the destination network gateway. For example, a destination IP address may be inserted into the header, the port number may be incremented by 2, and the packet is transmitted from the voice switch 585 to a destination gateway. As a result of this process, variations within H.323 networks are compensated for by the single voice switch 585 between the two networks.
The voice switch 585 requires some method of interpreting the information within the network type identifiers 610, 620 in order to properly translate addresses on the packets. According to one embodiment of the invention, a network type identifier table may be used. An example of a network type identifier table of network type identifiers is also shown in
(ii) Pre-Defined Port Range Representing a Network Type
A second method for identifying the network types within a VoIP connection provides that port numbers are assigned according to the two types of networks within the connection. This method begins at the call set-up stage during which the port numbers, on which packets will travel in a VoIP connection, are assigned. As with the first method, after both IP addresses of the two gateways are identified, they are compared and smaller and larger IP addresses are determined. A network pair table 700A is maintained within the voice connector 570 that relates port ranges to VoIP connections between network types. The use of a table, as opposed to a static division of the port number space into equal sized port ranges, allows allocation of larger port ranges to more common network type pairs and smaller port ranges to less common network type pairs. An example of such a network pair table 700A is illustrated in
During the set-up of the VoIP connection, the types of the networks are identified and a port number is assigned by the voice connector 570 according to these ranges defined within the network pair table 700A. This assigned port number is transmitted to the voice switch 585 and both gateways so that traffic between the two gateways may be forwarded correctly.
The network pair table 700A is also transmitted to the voice switch 585 if the voice switch 585 does not have the table 700A or the voice switch 585 has an old version of the table 700A. It is important for the voice switch 585 to have a current version of the table so that both gateway types may be properly identified from the port on which a packet arrives. As a result of this network pair table 700A, the voice switch 585 is able to identify the network types of both the larger and smaller IP addressed gateways by identifying the port range corresponding to the port used for packet transmission. For example, a VoIP connection is set-up between the first network gateway 212(a) and the second network gateway 222(a). Both gateways 212(a), 222(a) transmit packets to the same port, port N, of the voice switch 585. The voice switch 585 is able to identify the type of both gateways 212(a), 222(a) by comparing the port on which a packet arrives to the network pair table 700A shown in
Note that the network pair table 700A may be continually updated by the voice connector 570 simply through re-transmission to the voice switch 585. Also, the size of the network pair table 700A may be adjusted according to the number of different types of networks that use the voice switch 585 for VoIP connections. Also, the port ranges may be adjusted relative to the frequency of VoIP connections occurring between certain types of networks. For example, if VoIP connections between two types of networks occur very frequently, the port range corresponding to this connection may be increased to more efficiently accommodate these connections.
Adjustment of port ranges in the network pair table 700A requires the ability to dynamically allocate the port ranges. In contrast, a fixed-size implementation of the network pair table 700A does not allow for such adjustment, instead constraining the ranges of table 700A to the sizes initially set, e.g. at system startup. Dynamic allocation of port ranges provides more efficient assignment of port numbers than does static allocation, where port needs must be guessed at and fixed ahead of time. Dynamic allocation further allows one voice switch 585 to be shared by multiple voice connectors 570, or one voice connector 570 to use multiple voice switches 585, since assignment of the port ranges to the various voice switches can then be performed at runtime rather than requiring an estimate ahead of time. For example,
In one embodiment, such dynamic allocation is achieved by means of port range commands which one or more voice connectors 570 communicate to the port allocation module 599 of the voice switch 585 of
The range allocation command takes as arguments a desired number of ports and a requested lease time, e.g. specified as an integer number of seconds, during which the resulting port range will be considered allocated. The port allocation module 599 then checks the request, denying it responsive to factors such as the requested number of ports being unavailable or too large for a single request, and otherwise allocating the requested ports and returning an identifier and size of the allocated ports. For example, the port allocation module 599 could return more or fewer ports than requested, e.g. rounding to a port range size that is a power of 2. The returned identifier can then be used to reference that port range when issuing further port range commands, such as renewing or deallocating the port range. In one embodiment, the gateway types for the two networks are specified at the time that the port range is requested as part of the range allocation command; in other embodiments, they can be specified later, e.g. using a separate command.
The range deallocation command takes as an argument the port range identifier returned when the port range was initially allocated. The range renewal command likewise takes as an argument the port range identifier previously returned, and additionally takes a lease renewal time, e.g. a number of seconds, for which the port range will be considered reallocated.
In one embodiment, the port range commands are implemented using a common message format containing a set of fields representing the union of all the arguments used by the various commands, with the particular command specified by a command identifier field in the message format and only the relevant fields provided with specified values for any given message. For example, the range deallocation command requires a port range identifier for a port range to deallocate, and the range renewal command additionally requires a lease renewal time for which to renew the port range.
c) Packet Direction Identification
In addition to identifying the network types of both the first and second gateway 212(a), 222(a), the voice switch 585 identifies the direction a packet is traveling. This direction information allows the voice switch 585 to properly switch a packet to a correct destination address because both the first and second gateways 212(a), 222(a) are transmitting packets to the voice switch 585 on the same port (e.g., port N).
The buffer 920 comprises a first storage element 925 and a second storage element 927. The buffer 920 can implement a toggle for storing and for comparing the IP addresses. After receiving the source IP address from a packet that arrived on a particular port, the address comparator 910 compares this source address to the address within the first storage element 925. If this source IP address matches the IP address within the first storage element 925, the buffer 920 transmits the address in the second storage element 927 to the address comparator 910 via line 935, without the need to update the buffer. This address from the second storage element 927 is the destination address for the gateway in the connection to which the packet should be forwarded. This address is inserted into the header of the packet so that it may be forwarded to the correct gateway in the connection.
Comparatively, if the source IP address from a packet does not match the address in the first storage element 925, then the address in this first storage element 925 is transmitted back to the address comparator 910 via line 930. This address from the first storage element 925 is the destination address for the gateway in the connection to which the packet should be forwarded. The source IP address from the packet is stored within the first element 925 and the IP address that had previously been stored in the first storage element is transferred and stored in the second storage element 927. As a result, both IP addresses in the connection are continually stored within the buffer 920. The buffer 920, therefore, toggles the addresses when the source IP address does not match the address stored in the first storage element 925. If the buffer 920 is implemented as a stack, then the source IP address can be pushed onto the stack when there is no match in with the first storage element 925 (i.e., the top of the stack). In one embodiment, if the source IP address from the packet does not match either the address in the first storage element 925 nor the address in the second storage element 927, the address of the first storage element is moved to the second storage element and the source IP address is placed in the first storage element.
After a destination address is identified and a network type for both gateways has been determined, information is inserted into the packet header. For example, a new destination IP address and port number are inserted into the packet header. Thereafter, the packet is transmitted from voice switch 585 to a gateway (e.g., 212(a) or 222(a)) on a particular port.
d) Method for Translating a Network Address Within a Connection
The voice switch 585 identifies 1015 a direction of the packet or destination address to which the packet should be forwarded. This identification may be accomplished using numerous methods. For example, as described above, a buffer and comparator may be implemented whereby a destination address is determined using the source address within the packet header. Once both network type information and a packet direction have been determined, the voice switch 585 performs 1020 an appropriate address translation on the packet. Thereafter, the packet is transmitted 1025 to a correct gateway in the connection.
e) Gateway Behind Firewall
Thus, it is desirable for the voice switch 585 to acquire knowledge of the translated source address and port A5/P5 produced by the firewall 1110 based on the original address and port number A1, P1 of the network gateway 212(a). A process for doing so is illustrated in
The voice connector 570 provides this information to the voice switch 585 by sending 1210 a Connect command to the voice switch, in which it specifies the source address and port number extracted from the body of the connection request, along with an address mask and a port mask. In one embodiment, the port number includes both a port number for RTP and a port number for RTCP, and each has an associated mask. The masks allow the voice switch 585 to determine whether a given packet came from the firewall 1110 and thus to deliver the packet rather than rejecting it, as described below. The voice switch then stores 1215 the connection information specified in the Connect command for use with future voice packets.
When specifying the Connect command for delivery to the voice switch 585, the voice connector 570 can set the address and port masks to reflect the level of knowledge that it has about the firewall 1110. For example, if the voice connector 570 knows that there is no firewall 1110 between itself and the network gateway 212(a) (e.g., due to a match between the address information in the packet header and the body of a VoIP call setup packet), then it sets all the bits of the masks to 1, meaning that it will accept only the gateway address and port number themselves, and no other related addresses/port numbers. Similarly, if the voice connector 570 has no information at all about the firewall 1110 and wishes to allow communications to proceed, it sets all the bits of the masks to 0, indicating that all addresses and port numbers will be accepted. The more bits of the mask that are set to 1, the narrower the range of acceptable addresses, and hence the greater the degree of security provided. In one embodiment, the voice connector 570 obtains the value for the masks to be used for a given network gateway and port number from a configuration file, in cases where information on the firewall 1110 is static and known, and sets the value of the masks to 0 otherwise.
f) Method of Port Multiplexing
The technique described above for accounting for the presence of firewalls can also be extended to support port multiplexing. Port multiplexing, in which a single port is shared for a plurality of distinct communications, allows a reduction in the number of ports on which a device listens, and is useful for reasons such as network security. To achieve port multiplexing, the sender, such as the network gateway 212(a), replaces the original destination port number in a packet header, e.g. a UDP header, with a port number that the voice router 500 has designated as a multiplex port, and places the original destination port number at a known location in the body of the packet, e.g. as the last two bytes of the UDP payload. In one embodiment, the port number of the multiplex port is predefined; in another embodiment, the voice connector 570 sends a query to the voice switch 585 and in response receives a dynamically-allocated multiplex port number. When the voice switch 585 of the voice router 500 receives this packet, it examines the destination port number in the packet header, and if it is a port number known to represent a multiplex port, then the voice switch 585 extracts the original destination port number from the known location in the body of the packet and replaces the port number in the packet header with the original destination port number. The voice switch 585 then forwards the packet to the receiver, e.g., the network gateway 222(a). In one embodiment, the voice switch 585 identifies which port ranges represent multiplex ports by reading a configuration file.
While the present invention has been described with reference to certain preferred embodiments, those skilled in the art will recognize that various modifications may be provided. Variations upon and modifications to the preferred embodiments are provided for by the present invention, which is limited only by the following claims.
This application is a continuation-in-part of application Ser. No. 12/180,432 (attorney docket no. 14570), filed Jul. 25, 2008, from which priority is claimed under 35 U.S.C. §120. Application Ser. No. 12/180,432 is in turn a continuation of application Ser. No. 10/270,809 (attorney docket no. 06476, now U.S. Pat. No. 7,417,978), filed Oct. 14, 2002. Application Ser. No. 10/270,809 is in turn related to provisional application Ser. No. 60/338,077, filed Nov. 30, 2001, and provisional application Ser. No. 60/329,015, filed Oct. 12, 2001, from which it claims priority under 35 U.S.C. §119(e). Application Ser. Nos. 12/180,432, 10/270,809, 60/338,077, and 60/329,015 are hereby incorporated by reference.
Number | Date | Country | |
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60338077 | Nov 2001 | US | |
60329015 | Oct 2001 | US |
Number | Date | Country | |
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Parent | 10270809 | Oct 2002 | US |
Child | 12180432 | US |
Number | Date | Country | |
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Parent | 12180432 | Jul 2008 | US |
Child | 12908864 | US |