[asterisk-bugs] [JIRA] (ASTERISK-24330) Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24330?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233930#comment-233930 ] 

Matt Jordan commented on ASTERISK-24330:
----------------------------------------

Patch for 13/14/master up on Gerrit, tested with JSSIP:

*{{chan_sip}}:*

{code}
JsSIP:Transport received text message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS mv355e3cim2b.invalid;branch=z9hG4bK8985649;received=127.0.0.1
From: "webrtc" <sip:webrtc at 127.0.0.1:8089>;tag=c406nnt75p
To: <sip:webrtc at 127.0.0.1:8089>;tag=as34f4e40e
Call-ID: nke8scs73bopit8bp8f4r8
CSeq: 2 REGISTER
Server: Asterisk PBX GIT-13-13.4.0.0-1993-g845c5f4M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:u3gd5q2g at mv355e3cim2b.invalid;transport=ws>;expires=600
Date: Mon, 28 Nov 2016 19:40:40 GMT
Content-Length: 0

{code}

*{{res_pjsip}}:*

{code}
JsSIP:Transport received text message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS cg6elanq9ev8.invalid;rport=37609;received=127.0.0.1;branch=z9hG4bK5252818
Call-ID: od04q3nal4ppiqij2n86jq
From: "webrtc" <sip:webrtc at 127.0.0.1>;tag=u7uca7br4t
To: <sip:webrtc at 127.0.0.1>;tag=z9hG4bK5252818
CSeq: 2 REGISTER
Date: Mon, 28 Nov 2016 19:42:18 GMT
Contact: <sip:htqr4nmd at 127.0.0.1:37609;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX GIT-13-13.4.0.0-1993-g845c5f4M
Content-Length:  0

{code}

> Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
> -----------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24330
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24330
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/WebSocket, Resources/res_http_websocket
>    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
> {noformat}
> 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
> {noformat}
> 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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