[asterisk-bugs] [JIRA] Issue Comment Edited: (ASTERISK-20238) Registering SIP user from JavaScript sipML5 client fails when random ".invalid" domain sent in Contact header

[James Mortensen (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20238?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=195827#comment-195827 ] 

James Mortensen edited comment on ASTERISK-20238 at 8/15/12 6:16 PM:
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I've attached the DEBUG information from a call attempt. I can register with nat=yes option and the nat=yes, force_rport options.  However, the call attempt from sipML5 results in a 488 Status code.  

It sounds like it could be an encoding problem, but I'm not 100% sure.

      was (Author: james.mortensen):
    I've attached the DEBUG information from a call attempt. I can register with nat=on option and the nat=on, force_rport options.  However, the call attempt from sipML5 results in a 488 Status code.  

It sounds like it could be an encoding problem, but I'm not 100% sure.
  
> Registering SIP user from JavaScript sipML5 client fails when random ".invalid" domain sent in Contact header
> -------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20238
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20238
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.0.0-beta1
>            Reporter: James Mortensen
>         Attachments: sip_call_attempt.txt
>
>
> According to the spec, the JavaScript SIP client should send a random ".invalid" domain in the Contact header, since the JavaScript implementation has no way to determine the IP address or the port to send.
> It seems that the Asterisk server should instead use the existing port 8088 to send WebSocket responses back to the client; however, the end result is a 400 Bad Request.
> If I send my hard-coded IP address, I see a 200 OK message. I've also tried seeing the nat=no setting in sip.conf.

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