[asterisk-users] Asterisk now available with bundled pjproject!

[Jean-Denis Girard]

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:*91 at sysnux.pf SIP/2.0
Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368
Max-Forwards: 70
To: <sip:*91 at sysnux.pf>
From: <sip:websip2 at sysnux.pf>;tag=q1ejnhm074
Call-ID: l7rivm3clnebl6om63eb
CSeq: 1487 INVITE
Authorization: Digest algorithm=MD5, username="websip2",
realm="asterisk", nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6",
uri="sip:*91 at sysnux.pf", response="d30a2f2b4d5d25e81dded44b7d98e336",
opaque="639fdd14224f0290", qop=auth, cnonce="r0d44vjitbof", nc=00000001
Contact: <sip:cldsr32v at ca4cqpd5cv2h.invalid;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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