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I just install Kamailio 4.4.5 + RTPEngine on Ubuntu Server 16.04

all config copy from https://github.com/whisk/WEBRTC-to-SIP

And using SIP.js latest version to make call between 2 chrome browser.

SIP user register successfully and can using text chat But when User Invite another user any his Accept call, get this error

Failed to set remote answer sdp: Called with SDP without DTLS fingerprint.

How to solve this issue???

Hamid
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1 Answers1

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Use sdp_with_transport_like(...) instead of sdp_with_transport(...), because the transport string is UDP/TLS/RTP/SAVPF and the second function is doing an exact match, but the parameter in referred example is only RTP/SAVPF.

miconda
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