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I tried this suggestion https://github.com/OpenSIPS/opensips/issues/1996#issuecomment-644200253 to be able to play an early media when some one call to a fixed number. It's working well in local netwo…
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Hello.
I've looking into this issue in forum and github. Normally it happens after an error which is the cause, But,the only error I can see is not retrieving any image. What is happening? Hardware…
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Is there an easy way to call ForceKeyFrame method ?
So far the only way i see reaching this is:
1. Get senders & tracks from them
2. Create TrackContext from the track
3. Create encoder via NewR…
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I want to try to Implement nack by myself, but I encountered a problem
My current approach is
For receiver:
1. Call MediaStream.UseBuffer
2. Handle the OnRtpPacketReceived event
3. Check the …
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From [XEP-0167: Jingle RTP Sessions](https://xmpp.org/extensions/xep-0167.html); the document states that the \ is a childElement of \
However it seems that the childElement \ is always associated…
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# 0.1.0 (01.2024)
### Media reading/writing:
* [x] IVF reader/writer - #29 #44
* [x] Ogg reader/writer - #43 #50
* [x] VP8 RTP packetizer/depacketizer - #30 #45
* [x] Opus RTP packetize/depacketi…
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Hi guys,
I'm experiencing a big (for my use case) problem.
The recordings are quite often in desync with audio/video after a couple of minutes, sometimes it starts in desync.
I tried everything, last…
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## Issue description
aiortc is the WebRTC peer; when it sends SDP with per-media ice-ufrag/ice-pwd (i.e., m=audio, m=video have different ICE attributes) KMS will use the second pair (i.e. from m=vi…
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for example packetSent is represented by a 32-bit in RTP/RTCP. If it overflows does the webrtc-stats report:
1. keeps counting up, we are not limited in such cases we use "unsigned long long"
2. bec…
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_Last revised: Oct 2023_
## Overview
Create minimal DTLS encoder and decoder. See #229 for background.
DTLS works on transport level. Instead of sending RTP packets over UDP, we will pack RTP…