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Make docs smaller (#156)
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sgfn authored Aug 9, 2024
1 parent dc1ec48 commit 6d67f23
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6 changes: 3 additions & 3 deletions guides/advanced/debugging.md
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Expand Up @@ -94,12 +94,12 @@ Firstly, take a look at `chrome://webrtc-internals`. Make sure that there's noth
your PeerConnection is in the `connected` state. Find the `inbound-rtp` section (either table with stats or graphs, the graph section
will be called _Stats graphs for inbound-rtp (kind=video, mid=2, ...)_) related to your track.

![PeerConnection state](assets/state.png)
![PeerConnection state](assets/state.jpg)

If you cannot find the `inbound-rtp` section, make sure you properly added and negotiated the tracks. The SDP offer and answer can be inspected in the API trace section of
`chrome://webrtc-internals`.

![InboundRTP](assets/inbound.png)
![InboundRTP](assets/inbound.jpg)

Assuming you have found your `inbound-rtp` section, take a look at `packetsReceived` graph.

Expand All @@ -116,7 +116,7 @@ transceiver will drop all of the incoming packets. For instance, creating a `rec
direction (not `sendonly`!) on the remote peer, which might be counterintuitive. Make sure you properly negotiated the session by inspecting the SDP offers and answers.
If you're sure that the session was properly negotiated, reproduce the issue, create an RTP dump, and share it with us - this might be an Elixir WebRTC bug.

![CandidatePair](assets/pair.png)
![CandidatePair](assets/pair.jpg)

#### 2. `packetsReceived` growing, but `framesDecoded` stays on 0 (in case of video)

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2 changes: 1 addition & 1 deletion lib/ex_webrtc/rtp/depayloader.ex
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Expand Up @@ -2,7 +2,7 @@ defmodule ExWebRTC.RTP.Depayloader do
@moduledoc """
RTP depayloader.
It unpacks RTP pakcets into audio/video frames.
It unpacks RTP packets into audio/video frames.
"""

alias ExWebRTC.RTPCodecParameters
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