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Set option sendrpid to no for webrtc line.

username-removed-183524 requested to merge 657-fix-webrtc-hangup-after-xfer into master

The reason is that re-INVITE from asterisk to update RPID (or PAI) header has the SDP directive setup set to active. SIP ML5 expects the SDP directive setup set to actpass on INVITE method.

Seems to be correct according to RFC 5763 (DTLS-SRTP) in section 5.

Closes #657

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