I am trying to create a conference feature in our webRTC based softphone. I believe I have the proper audio mixing on the client end, but when I renegotiate, I see an Rx buffer overflow error:
[Dec 9 22:35:32] ERROR: pjproject: <?>: sip_endpoint.c Error processing packet from 127.0.0.1:48930: Rx buffer overflow (PJSIP_ERXOVERFLOW) [code 171062]: INVITE sip:127.0.1.1:8089;transport=ws SIP/2.0 Via: SIP/2.0/WSS bor70uj1f6ac.invalid;branch=z9hG4bK6327172 Max-Forwards: 69 To: <sip:2102@domain>;tag=8c99ed9d-3b74-4031-a263-e4fe0bc5d479 From: "Name" <sip:ext@domain>;tag=6146n3nqd4 Call-ID: vpeifvndt4he93ost25j CSeq: 9776 INVITE Contact: <sip:firstname.lastname@example.org;transport=ws;ob> Content-Type: application/sdp Session-Expires: 90;refresher=uac Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: timer,ice,replaces,outbound User-Agent: JsSIP 3.3.6 Content-Length: 3625
I found this stack overflow article with the same issue: https://stackoverflow.com/questions/39275328/asterisk-13-10-pjsip-webrtc-rx-buffer-overflow-pjsip-erxoverflow
The problem with this solution is we do not build asterisk or pjsip from source, so we can not overwrite the constants that the selected answer suggests.
My question is there another way to get around this or should I just find some way of using the ConfBridge application instead?