ZyXEL UAG715 User Guide - Page 215

ALG and Trunks, H.323

Page 215 highlights

Chapter 18 ALG port numbers and allows the related sessions to go through the firewall so the application's traffic can come in from the WAN to the LAN. ALG and Trunks If you send your ALG-managed traffic through an interface trunk and all of the interfaces are set to active, you can configure routing policies to specify which interface the ALG-managed traffic uses. You could also have a trunk with one interface set to active and a second interface set to passive. The UAG does not automatically change ALG-managed connections to the second (passive) interface when the active interface's connection goes down. When the active interface's connection fails, the client needs to re-initialize the connection through the second interface (that was set to passive) in order to have the connection go through the second interface. VoIP clients usually reregister automatically at set intervals or the users can manually force them to re-register. FTP File Transfer Protocol (FTP) is an Internet file transfer service that operates on the Internet and over TCP/IP networks. A system running the FTP server accepts commands from a system running an FTP client. The service allows users to send commands to the server for uploading and downloading files. H.323 H.323 is a standard teleconferencing protocol suite that provides audio, data and video conferencing. It allows for real-time point-to-point and multipoint communication between client computers over a packet-based network that does not provide a guaranteed quality of service. NetMeeting uses H.323. SIP The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol that handles the setting up, altering and tearing down of voice and multimedia sessions over the Internet. SIP is used in VoIP (Voice over IP), the sending of voice signals over the Internet Protocol. SIP signaling is separate from the media for which it handles sessions. The media that is exchanged during the session can use a different path from that of the signaling. SIP handles telephone calls and can interface with traditional circuit-switched telephone networks. RTP When you make a VoIP call using H.323 or SIP, the RTP (Real time Transport Protocol) is used to handle voice data transfer. See RFC 1889 for details on RTP. UAG715 User's Guide 215

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Chapter 18 ALG
UAG715 User’s Guide
215
port numbers and allows the related sessions to go through the firewall so the application’s traffic
can come in from the WAN to the LAN.
ALG and Trunks
If you send your ALG-managed traffic through an interface trunk and all of the interfaces are set to
active, you can configure routing policies to specify which interface the ALG-managed traffic uses.
You could also have a trunk with one interface set to active and a second interface set to passive.
The UAG does not automatically change ALG-managed connections to the second (passive)
interface when the active interface’s connection goes down. When the active interface’s connection
fails, the client needs to re-initialize the connection through the second interface (that was set to
passive) in order to have the connection go through the second interface. VoIP clients usually re-
register automatically at set intervals or the users can manually force them to re-register.
FTP
File Transfer Protocol (FTP) is an Internet file transfer service that operates on the Internet and over
TCP/IP networks. A system running the FTP server accepts commands from a system running an
FTP client. The service allows users to send commands to the server for uploading and downloading
files.
H.323
H.323 is a standard teleconferencing protocol suite that provides audio, data and video
conferencing. It allows for real-time point-to-point and multipoint communication between client
computers over a packet-based network that does not provide a guaranteed quality of service.
NetMeeting uses H.323.
SIP
The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol that handles
the setting up, altering and tearing down of voice and multimedia sessions over the Internet. SIP is
used in VoIP (Voice over IP), the sending of voice signals over the Internet Protocol.
SIP signaling is separate from the media for which it handles sessions. The media that is exchanged
during the session can use a different path from that of the signaling. SIP handles telephone calls
and can interface with traditional circuit-switched telephone networks.
RTP
When you make a VoIP call using H.323 or SIP, the RTP (Real time Transport Protocol) is used to
handle voice data transfer. See RFC 1889 for details on RTP.