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objective c - AC3 audio FramesPerPacket size - Stack …
https://stackoverflow.com/questions/11704351/ac3-audio-framesperpacket-size
1. This answer is not useful. Show activity on this post. The AC3 frame size should be 256 * channels, thus here I would expect: mFramesPerPacket = 256 * mChannelsPerFrame. Share. Improve this answer. Follow this answer to receive notifications. answered Aug 15 '12 at …
Audio Packet Format - Poly Community
https://community.polycom.com/polycom/attachments/polycom/VoIP/7701/2/ea70568-audio-packet-format.pdf
2 G.722 – Typical audio payload is 240 bytes (30ms) 3 G.711u – Typical audio payload is 240 bytes (30ms) Audio data consists of a 6 byte audio header followed by two frames of audio data. The first frame is a redundant frame—it contains a copy of the audio from the previous packet. The second frame contains the current audio.
Troubleshooting Tip: One way Audio issue in VOIP c ...
https://community.fortinet.com/t5/FortiGate/Troubleshooting-Tip-One-way-Audio-issue-in-VOIP-calls-caused-by/ta-p/203336
Check for IP and Audio/RTP port. Connection information changing (IP/Port changing) will be visible if traffic is processed by SIP ALG. Below are the sample packets as an example. Packet Capture on Inside Interface: Frame 44: 863 bytes on wire (6904 bits), 863 …
SIP Protocol: What Is & How It Works in a VOIP Call ...
https://www.softwareadvice.com/resources/what-is-sip/
Encoded packets of audio data are carried by the real-time transport protocol (RTP), a specialized application layer protocol used for real-time streaming of audio and video data. RTP sessions are independent of SIP. RTP sessions run parallel to SIP sessions, unlike SDP, which is a payload of SIP. ...
Record-A-Call
https://edocs.mitelcommunicationservice.com/appsuitelatest/Record_A_Call.html
Set the Audio Frames/IP Packet to 2 Maximum Number Of Ports -> Set the maximum number of ports to be the same as the number of calls you wish to be recorded, the maximum supported at this time is 8. Call Failure Threshold -> 500 (Set this as high as possible to reduce the risk of calls not being recorded).
c++ - FFmpeg av_read_frame returns packets from audio ...
https://stackoverflow.com/questions/51595363/ffmpeg-av-read-frame-returns-packets-from-audio-stream
My issue is that the video stream does not match the one from the packet returned by av_read_frame. The video stream is obtained looping on the available streams until the video stream is found. Then when retrieving the frame data, it seams grabbing the audio channel. while (av_read_frame (pFormatCtx, &packet) >= 0) { // read returns 0 // Is ...
Encrypted vs Non-Encrypted SIP Packets in Wireshark ...
https://www.nurango.ca/blog/encrypting-voip-comparing-in-wireshark
An admin decides to encrypt the SIP packets but not the audio – A malicious network user can now sniff out the audio packets from all of your conversations and play them back. Just as bad – the attacker can also capture DTMF (touch tone) sounds over the network and capture credit card and account data.
How to determine audio "frame rate" : audioengineering
https://www.reddit.com/r/audioengineering/comments/dv9oab/how_to_determine_audio_frame_rate/
Audio has no frame rate. It only has a sample rate. For video, this is almost always 48 kHz, but sometimes 44.1. The sample rate is in no way connected to the video frame rate. The important thing for audio sync is that both sound and picture are being played back at the exact same speed they were recorded.
VoIP Basics: Codec Latency vs. Bandwidth Optimization
http://www.toncar.cz/Tutorials/VoIP/VoIP_Basics_Codec_Latency_vs_Bandwidth.html
VoIP Basics: Codec Latency vs. Bandwidth Optimization. As we have shown in the overview of codecs, the low-bandwidth codecs are quite efficient. For example, G.729 will compress 10 milliseconds of audio to 10 bytes and G.723.1 encodes 30ms frames to 24 or 20 bytes. However, since we send compressed audio frames as payload in RTP packets which ...
Jumbo Frames - Networking, Networked Audio, and …
https://audiophilestyle.com/forums/topic/59460-jumbo-frames/
frames would be forced to segment the IP packet into multiple frames I doubt that audio server applications use a larger packet size because the files aren't that large and because you typically buffer only small amount of data before starting play. The trade off with larger packet/frame size is greater inefficiency on lost/errored packets.
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