[asterisk-bugs] [JIRA] (ASTERISK-23191) Cannot get local SDP on incoming calls (WebRTC)

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23191?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214831#comment-214831 ] 

Rusty Newton edited comment on ASTERISK-23191 at 2/4/14 5:31 PM:
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First, you have users.conf options in your sip.conf that won't do anything. Hopefully they are not causing issues. That is, *hasiax*,*hassip*,*autoprov* are all users.conf options. Remove those from your sip.conf configuration.

Second, you missing the "t" in "direcmedia=no" for peer 1060. That could definitely cause unexpected behavior for you.

Third, you have "allow=all" for your codec definitions, which almost never makes sense when you know what is on the other end. I also recently saw some strangeness with this. Why would you want Asterisk to offer and potentially accept a large variety of codecs when you know only one or two are supported by the far end?

Please fix the other issues and try 
{noformat}
disallow=all
allow=ulaw
{noformat}
For your codec settings.

This looks to be a support issue and possibly not a bug, plus your whole configuration depends on patches that are not included yet (and could have bugs in themselves). If those fixes don't resolve your issue, then we'll go ahead and close out the issue pending inclusion of the patches you are using first. We need to fix one thing at a time.
                
      was (Author: rnewton):
    First, you have users.conf options in your sip.conf that won't do anything. Hopefully they are not causing issues. That is, *hasiax*,*hassip*,*autoprov* are all users.conf options. Remove those from your sip.conf configuration.

Second, you missing the "t" in "direcmedia=no" for peer 1060. That could definitely cause unexpected behavior for you.

Third, you have "allow=all" for your codec definitions, which almost never makes sense when you know what is on the other end. I also recently saw some strangeness with this. Why would you want Asterisk to offer a large variety of codecs when you know only one or two are supported by the far end?

Please fix the other issues and try 
{noformat}
disallow=all
allow=ulaw
{noformat}
For your codec settings.

This looks to be a support issue and possibly not a bug, plus your whole configuration depends on patches that are not included yet (and could have bugs in themselves). If those fixes don't resolve your issue, then we'll go ahead and close out the issue pending inclusion of the patches you are using first. We need to fix one thing at a time.
                  
> Cannot get local SDP on incoming calls (WebRTC)
> -----------------------------------------------
>
>                 Key: ASTERISK-23191
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23191
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_http_websocket
>    Affects Versions: 11.7.0
>            Reporter: Jay Jideliov
>            Assignee: Matt Jordan
>         Attachments: configs.txt, debuglog-cleared-ip.txt, Dump.txt.txt
>
>
> This is an issue created as a follow-up to this comment (https://issues.asterisk.org/jira/browse/ASTERISK-21930?focusedCommentId=211742&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-211742) as well as due to us having the same issue when trying to receive incoming calls via WebRTC.
> Asterisk 11.7 + SIPML5 + WSS, both Firefox and Chrome.
> Devices register fine, and are able to make outgoing calls. However, the issue arises on incoming calls, when all we get is the error (cannot get local sdp) after which the line disconnects.

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