[asterisk-bugs] [JIRA] (ASTERISK-24330) SIPoWS in Asterisk fill transport=wss which is'nt correct
cervajs (JIRA)
noreply at issues.asterisk.org
Tue Sep 16 12:36:29 CDT 2014
cervajs created ASTERISK-24330:
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Summary: SIPoWS in Asterisk fill transport=wss which is'nt correct
Key: ASTERISK-24330
URL: https://issues.asterisk.org/jira/browse/ASTERISK-24330
Project: Asterisk
Issue Type: Bug
Security Level: None
Affects Versions: 11.12.0
Reporter: cervajs
i'm trying make webrtc client with SIP.js library
i have working asterisk 11.12.0 wss configuration
when i REGISTER the header looks like
REGISTER sip:sip.example.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.136;branch=z9hG4bK7133687
Max-Forwards: 70
To: <sip:web_101_5416b222f0efa at sip.example.com>
From: <sip:web_101_5416b222f0efa at sip.example.com>;tag=qok8o8thta
Call-ID: 1np04l3k9bumr5aqbi6tpr
CSeq: 81 REGISTER
Contact: <sip:rlr4g88u at 192.0.2.136;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:f55544fa-c05d-468b-9c51-fd7a837864b5>";expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
Supported: path,gruu,outbound
User-Agent: SIP.js/0.6.2
Content-Length: 0
the problem is in the contact line
Contact: <sip:rlr4g88u at 192.0.2.136;transport=ws>;
it's correct - http://tools.ietf.org/html/rfc7118#section-5.2
but it's not working.
if i change the code in SIP.js to transport=wss then it works
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