RTP Port Devices Driver

Recommended Firewall Settings

The rtpMIDI driver is a network MIDI-driver for Windows 7 up to Windows 10. It can be installed and used on both 32bit and 64bit systems. Every Apple-computer based on OS X since “Tiger” (10.4) includes a network-MIDI driver as a standard system-component. These downloads cover most Intel® Ethernet Adapters and install the latest drivers when you run them. The network adapter property sheet in Windows® 10 provides information about the network adapter and driver on the computer. Follow these steps to open the adapter properties: Right-click the Start button. Click Device Manager from the list. Download the latest drivers, firmware, and software for your HP ENVY 4520 All-in-One Printer.This is HP’s official website that will help automatically detect and download the correct drivers free of cost for your HP Computing and Printing products for Windows and Mac operating system. A RTP-port-number specify the audio RTP port number (for multicast streaming only)-A specify multicast streaming (to the given multicast address)-aac encode and stream audio as MPEG-4 AAC, with the specified bitrate -amr: encode and stream audio as AMR, with the specified bitrate -brightness. If a device is disabled, it has a red X across its icon, like the Bluetooth Communications Port in this picture. To enable it, right click on the device and select “Enable.” Afterwards, the device should not have a red X across its icon. Devices can also have issues loading or have device driver problems.

Rtp port devices driver license test

Modified on: Fri, 4 Oct, 2019 at 12:24 PM


T38Fax owns (AS7324) and (AS396431), and all of our services operate out of these address blocks. We use SIP over UDP for call signaling. For outbound calling and registration via SIP, you can either use the standard UDP port 5060 or the nonstandard UDP port 5080. Use of the non-standard SIP port 5080 is recommended to avoid SIP ALGs: this is discussed in more detail here. For inbound calls, we will attempt to establish the call from port 5060, unless you are both using SIP registration and sending your registration to 5080. When both of these conditions are met, we will also send inbound calls from port 5080. The other ports referenced are for RTP or UDPTL: the media streams. Note that in a single session, the SIP packets will often flow between an IP address different from that of the RTP packets.

ACL Rules

For simplicity, some customers may wish to whitelist the and ranges in their firewall or fail2ban, as doing so allows all traffic described above in a single firewall rule. Administratively opening ports, especially SIP ports, to receive traffic from any and all IPs is not recommended. Alternatively, if you would like to use the most specific ruleset possible, allow only the traffic from below:

Port Forwarding

When using registration authentication, most devices do not require any port forwarding to work with our service. If you use IP authentication, you will need to forward your SIP port: often UDP port 5060, 5160, or 5080, depending on which port your SIP driver is listening. If you are using an Asterisk-based PBX, please also note the port forwarding requirements mentioned in the Asterisk Design Guide.

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'wis-streamer' is a standards-compliant RTSP/RTP server applicationthat streams encoded video and audiofrom the LinuxWIS GO7007 driver.This driver supports several low-cost, off-the-shelf hardware video encoder devices,including:
  • PlextorConvertX TV402U (also EUand JP models)
The 'wis-streamer' applicationcan stream video in MPEG-4 (default), MPEG-1, MPEG-2 or motion-JPEG format,and can stream audio in raw PCM, 8-bit u-law, or MPEG-2 audio format.(It can also, as an option, stream MPEG-2 video and audio in a combinedMPEG Transport Stream.)

For open source software for streaming from other hardware encoders - or frompre-encoded files - see the'LIVE555 Streaming Media' software.

Building 'wis-streamer' from source code

Note: This application currently requires a Linux version 2.6 (or greater) kernel.
  1. First, download, build and install theWIS GO7007 driver.
  2. Download and build the'LIVE555 Streaming Media' libraries.
  3. Download the'wis-streamer' source code package,and store it in the same directory as the 'live' directory that you unpackedand built in the previous step. (I.e., don't store it in the'live' directory, but store it in the directory that contains'live'.)
  4. Run:

Downloading a pre-built 'wis-streamer' binary

Rtp port devices driver license testApre-built 'wis-streamer' executable binaryfor Linux/x86 is also available.This binary was built for 'Fedora' Linux kernel version 2.6.9, and may or may not workon other Linux versions.No support is provided for this binary; if it doesn't work for you, then build the applicationfrom source instead.

RTP Port Devices Driver

Running 'wis-streamer'

Runand the server will initialize with default parameters:unicast streaming;MPEG-4 video (640x480, up to 1.5 Mbps);raw PCM audio (16 bit, 48000 Hz);RTSP server port 8554;assuming a NTSC input source, on input device number 0 (composite video).(See belowfor how to change these parameters.)If the server initializes correctly, it will then print out a'rtsp://' URL that can be used - by a media player client - to play the stream.

Playing the stream

The stream can be played by a standards-compliant RTSP/RTP media player application.The VLC media player is recommended.On Windows and Mac OS, you can also runQuickTime Player.(Also, the'openRTSP'command-line application can be used to examine the RTSP protocol exchange,and/or record the stream's audio and video data.)


For questions about the'LIVE555 Streaming Media' libraries,and/or the 'wis-streamer' source code,use thedevelopers' mailing list:'[email protected]'.(Note that you must subscribe to this mailing list before you can post to it.)

Note, however, that no support is currently provided for theWIS GO70007 encoder driver; please refer to the source code.

Copyright Notice

Note that the 'wis-streamer' application uses the'LIVE555 Streaming Media' libraries,whose source code is licensed under the GNU LGPL.Therefore, the 'wis-streamer' application is a 'Combined Work' that issubject to the conditions of the LGPL,even though the 'wis-streamer' source code itself is not licensed under the LGPL.For more information about your obligations under the LGPL, seehere.

Command-line options

Input device selection

When 'wis-streamer' initializes, it prints out the available input devices.By default, 'wis-streamer' reads from 'composite input' (device number 0).To have it read instead from another input device, use the'-i <input-device-number>' option.

By default, the input video source is assumed to be in NTSC format.For other formats, you can specify the input video type using the'-t <video-input-type>'option, e.g. '-t pal'.(To see a list of valid video input types, use '-t help'.)

If the selected input device is a TV tuner, then add the option'-c <band-name>:<channel-name>'to select a TV channel.(To see a list of valid <band-name>s, use'-c help'.To see a list of valid <channel-name>s for a given<band-name>, use'-c <band-name>:help'.)

Video output format

The video encoding format can be specified using one of options'-mpeg1','-mpeg2','-mpeg4' (the default),or'

Rtp Port Devices Driver Updater

-mjpeg'.Alternatively, use the '-nv' (i.e., 'no video') option to disable video encoding/streaming entirely.

To specify the width and height - in pixels - of the captured video, use the'-w <width>'and'-h <height>'options.(The default values are: <width>=640; <height>=480.)

The'-r <bitrate>'optioncan be used to specify a maximum video bitrate (in bits-per-second).(The default maximum video bitrate is 1500000 bps - i.e, 1.5 Mbps.)

The'-R <frame-rate>'option can be used to set the video frame rate (in frames-per-second) generated bythe encoder hardware. The <frame-rate> parameter can be either apositive integer, or a fraction of the form <numerator>/<denominator>(where <numerator> and <denominator> are bothpositive integers). The default frame rate is 30000/1001 fps for NTSC, and 25 fps for PAL or SECAM.When it initializes, 'wis-streamer' will display the actual frame rate generated by the encoder(which might not be exactly the same as the value that was asked for).

Audio output format

The'-f <sampling-frequency>'optionspecifies the audio sampling frequency (in Hz).The default value is 48000 Hz.

The streamed audio format can be specified using one of the following options:

  • '-pcm'which specifies raw, 16-bit PCM (the default).(Therefore, the resulting audio bitrate is<sampling-frequency>×2×16 bps.)
  • '-ulaw'which specifies 8-bit µ-law PCM.(Therefore, the resulting audio bitrate is<sampling-frequency>×2×8 bps.)
  • '-mpegaudio <bitrate-in-kpbs>'which encodes and streams MPEG-2 audio, with the specified bitrate- e.g., '-mpegaudio 128'. (Unlike video encoding, which is performed by the WIS encoder hardware,this MPEG audio encoding is done in software, in the 'wis-streamer' application itself.)
  • '-aac <bitrate-in-kpbs>'which encodes and streams MPEG-4 AAC audio, with the specified bitrate- e.g., '-aac 128'. (Unlike video encoding, which is performed by the WIS encoder hardware,this MPEG audio encoding is done in software, in the 'wis-streamer' application itself.)
  • '-amr'which encodes and streams AMR audio.(Unlike video encoding, which is performed by the WIS encoder hardware,this AMR audio encoding is done in software, in the 'wis-streamer' application itself.)
  • '-na' (i.e., 'no audio'), which disables audio entirely.

By default, the server's audio stream is stereo.To stream in mono instead (thereby halving the audio bitrate),use the'-M'option.

Special notes:

Rtp Port Devices Driver Windows 7

  • QuickTime Player has a longstanding bug that prevents it from playingµ-law audio (except for the special case of 8000 Hz mono).
  • Currently, if AMR audio encoding is requested, then the encoding will be mono only (regardless of whether or not the '-M' option is used).
  • There is currently a bug in AAC audio encoding, if mono (the '-M' option) is requested. (Mono AAC audio encoding currently does not work correctly.)

MPEG Transport streaming

As a special case, instead of using the video and audio format optionsnoted above, you can use the'-mpegtransport <audio-bitrate-in-kpbs>'option to specify that the streamed data will be a MPEG Transport Stream,containing MPEG-2 video, and MPEG-2 audio (with the specified bitrate).(If the <audio-bitrate-in-kpbs> parameter is zero, then no audiowill be included in the Transport Stream.)

Unicast or multicast streaming

By default, the server streams via unicast.This means that each request (from a client media player) to play the streamcauses a new stream to be transmitted, directly to the client.Note, therefore, that the maximum possible number of concurrent clients is limited by yournetwork bandwidth.

Alternatively, if the'-m' option is given, then the server will stream via IP multicast.This means that the server transmits the stream only once, regardless of how manyclients have asked to play it.The benefit of IP multicast is that it allows a potentially unlimited number of concurrentclients.The drawback of IP multicast, however, is that most routers do not support multicastby default,so that - in many cases - multicast streaming will not work outside a single LAN.

Special note:

By default, the server usessource-specific multicast (SSM),using a randomly-chosen SSM IP multicast address.To specify a different multicast address, use the'-A <multicast-address>' option(where <multicast-address> is an IPv4 multicast address, in'dotted quad' form; IPv6 is not yet supported).If the '-A' option is used without '-m', then 'any-source multicast (ASM)'is used instead of SSM.

By default, when streaming via multicast, the server uses RTP port number 6000 for video,and 6002 for audio.These port numbers can, however, be changed using the'-v <video-RTP-port-number>'and/or'-a <audio-RTP-port-number>'numbers.(When streaming via unicast, arbitrary (ephemeral) RTP port numbers are used.)

Client user authentication

DevicesBy default, there are no restrictions on who can request to play the stream.However, one or more'-u <username>:<password>'options can be used to specify valid username-password pairs.If this option is used, then the media player client must enter a valid username and password before it can play the stream.

Special note:

  • This feature does not cause the audio/video RTP data to be encrypted.
  • If you're using the VLC media player client, then the username and password must beentered in the prefix of the 'rtsp://' URL - i.e.,(Note that this causes the username and password to be passed over the network in the clear.)

Miscellaneous options

  • The'-brightness <value>','-contrast <value>','-saturation <value>',and'-hue <value>'options can be used to control the video input.If these options are omitted, default values are used.(To see the range of possible values, and the default value,use 'help' for <value>.)
  • By default, the server uses (TCP) port number 8554 for RTSP.This, however, can be changed using the'-p <RTSP-port-number>'option.

    Special note:

    If you run the server as 'root', you can use '-p 554', and the port number will be omitted fromthe 'rtsp://' URL (because 554 is the default port number in 'rtsp://' URLs).
  • The'-D <stream-description-string>'option can be used to set the textual stream description that gets passed(within a SDP description) to each RTSP client.Some media player clients may choose to display this string (e.g., in a banner).

Summary of command-line options

-a <audio-RTP-port-number>specify the audio RTP port number (for multicast streaming only)
-A <multicast-address>specify multicast streaming (to the given multicast address)
-aac <bitrate-in-kbps>encode and stream audio as MPEG-4 AAC, with the specified bitrate
-amrencode and stream audio as AMR, with the specified bitrate
-brightness <value>specify the video input brightness
-c <band-name>:<channel-name>specify a TV channel (for a TV tuner input device)
-contrast <value>specify the video input contrast
-D <stream-description-string>specify the stream's textual description (for SDP)
-f <sampling-frequency>specify the audio sampling frequency (in Hz)[default: 48000 Hz]
-h <height>specify the video image height (in pixels)[default: 480]
-hue <value>specify the video input hue
-i <input-device-number>specify the input device number (0: composite video[default]; 1: S-Video; 2: tuner (if present))
-mspecify multicast streaming (using source-specific multicast)
-Mstream audio in mono
-mjpegencode and stream video as motion-JPEG
-mpeg1encode and stream video as MPEG-1
-mpeg2encode and stream video as MPEG-2
-mpeg4encode and stream video as MPEG-4[default]
-mpegaudio <bitrate-in-kbps>encode and stream audio as MPEG-2, with the specified bitrate
-mpegtransport <audio-bitrate-in-kbps>encode and stream video and audio as MPEG-2 (with the specified audio bitrate), combined into a MPEG Transport Stream
-nadon't stream audio
-nvdon't stream video
-p <RTSP-port-number>specify the RTSP server port number[default: 8554]
-pcmstream audio as raw, 16-bit PCM[default]
-r <bitrate>specify the maximum streamed video bitrate (in bits-per-second)[default: 1500000 bps]
-R <frame-rate>specify the video frame rate (in frames-per-second)[default: 30000/1001 fps for NTSC; 25 fps for PAL or SECAM]
-saturation <value>specify the video input hue
-t <video-input-type>specify the video input type[default: ntsc]
-u <username>:<password>specify a username-password pair for client authentication
-ulawstream audio as 8-bit µ-law PCM
-v <video-RTP-port-number>specify the video RTP port number (for multicast streaming only)
-w <width>specify the video image width (in pixels)[default: 640]

Live Networks, Inc. (LIVE555.COM)

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