Basado en la librería SIPML5, combinando las capacidades de WebRTC y el protocolo SIP para ofrecer comunicaciones de voz y video directamente desde un navegador web compatible.
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Problem in BYE send from SIPML5 - Webrtc2sip (Version 2.6.0) #225
What steps will reproduce the problem?
1. Make a Call from Sip Client (softphone registered on a PBX)
2. Call is Answer in SIPML5, next Hangup the Call in SipML5
3. Caller (Softphone) not drop the call because an Error in BYE
What is the expected output? What do you see instead?
Softphone and Sipml5 drops the call.
What version of the product are you using? On what operating system?
Webrtc2sip VERSION: 2.6.0 in Centos 6.4
Please provide server logs with DEBUG level equal to INFO
fragment of Wireshark capture:
1 Request: BYE
sip:XXX.XXX.140.140:5061;transport=tcp;gsid=838d6170-e2e0-11e4-a45f-b4b52f6a27d4
2 Status: 407 Proxy Authentication Required
3 Request: BYE
sip:XXX.XXX.140.140:5061;transport=tcp;gsid=838d6170-e2e0-11e4-a45f-b4b52f6a27d4
4 Status: 403 Forbidden (Unauthorized)
Detail of Proxy-Authorization Header in Bye send from Webrtc2sip (line 3),
error in Username, "webrtc2sip" in not my username, username is 11001:
Digest
username=\"webrtc2sip\",realm=\"mypbx.com\",nonce=\"14cb9826a095c3d82caa2ffbfc40
8dda3723ac56fac\",uri=\"sip:XXX.XXX.140.140:5061;transport=tcp;gsid=838d6170-e2e
0-11e4-a45f-b4b52f6a27d4\",response=\"9bc020fdc826723d052639974dc30ff5\",algorit
hm=MD5,cnonce=\"0d6d6eb700ef0bdc2c71adb5e228144d\",opaque=\"1234567890abcedef\",
qop=auth,nc=00000001
Original issue reported on code.google.com by lperezu1...@gmail.com on 12 May 2015 at 9:00
Original issue reported on code.google.com by
lperezu1...@gmail.com
on 12 May 2015 at 9:00Attachments: