WebRTC Softphone: Rx buffer overflow (PJSIP_ERXOVERFLOW)

Hello there!

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[23244]: pjproject: <?>: 	        sip_endpoint.c Error processing packet from Rx buffer overflow (PJSIP_ERXOVERFLOW)  [code 171062]:
INVITE sip:;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:bq2g3151@bor70uj1f6ac.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90;refresher=uac
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?

There is no other way. To support large SDPs constants have to be adjusted, there is no working around that.

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.