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WEBRTC_SET_REMOTE_DESCRIPTION_ERROR: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.
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Your server is not sending a DTLS fingerprint inside the SDP when inviting the sip_ua client to start a call.
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WebRTC uses encryption by Default, all WebRTC communications (audio, video, and data) are encrypted using DTLS and SRTP, ensuring secure communication. Your PBX must be configured to use DTLS/SRTP when calling sip_ua.
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## Why isn't there a UDP connection option?
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This package uses a WS or TCP connection for the signalling processs to initiate or terminate a session (sip messages).
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Once the session is connected WebRTC transmits the actual media (audio/video) over UDP.
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If anyone actually still wants to use UDP for the signalling process, feel free to submit a PR with the large amount of work needed to set it up, packet order checking, error checking, reliability timeouts, flow control, security etc etc.
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## SIP/2.0 488 Not acceptable here
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The codecs on your PBX server don't match the codecs used by WebRTC
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-**opus** (payload type 111, 48kHz, 2 channels)
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-**red** (payload type 63, 48kHz, 2 channels)
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-**G722** (payload type 9, 8kHz, 1 channel)
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-**ILBC** (payload type 102, 8kHz, 1 channel)
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-**PCMU** (payload type 0, 8kHz, 1 channel)
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-**PCMA** (payload type 8, 8kHz, 1 channel)
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-**CN** (payload type 13, 8kHz, 1 channel)
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-**telephone-event** (payload type 110, 48kHz, 1 channel for wideband, 8000Hz, 1 channel for narrowband)
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</details>
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## NOTE
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Thanks to the original authors of [JsSIP](https://github.com/versatica/JsSIP) for providing the JS version, which makes it possible to port the [dart-lang](https://dart.dev).
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-[José Luis Millán](https://github.com/jmillan)
@@ -69,6 +107,7 @@ The project is inseparable from the contributors of the community.
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