Hi George, Le 07/03/2016 12:53, George Joseph a écrit : > Le 07/03/2016 09:28, George Joseph a écrit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released.
I don't think this is related to the bundled version, but I got PJSIP_ERXOVERFLOW when initiating a WebRTC video call from Chrome: [Mar 12 19:08:37] ERROR[9071]: pjproject:0 <?>: sip_endpoint.c Error processing packet from 192.168.10.88:50072: Rx buffer overflow (PJSIP_ERXOVERFLOW) [code 171062]: INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368 Max-Forwards: 70 To: <sip:*[email protected]> From: <sip:[email protected]>;tag=q1ejnhm074 Call-ID: l7rivm3clnebl6om63eb CSeq: 1487 INVITE Authorization: Digest algorithm=MD5, username="websip2", realm="asterisk", nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6", uri="sip:*[email protected]", response="d30a2f2b4d5d25e81dded44b7d98e336", opaque="639fdd14224f0290", qop=auth, cnonce="r0d44vjitbof", nc=00000001 Contact: <sip:[email protected];transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.7.3 Content-Length: 3335 ... This can be solved by adding the following line to config_site.h: #define PJSIP_MAX_PKT_LEN 6000 Would you consider adding it? Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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