[asterisk-bugs] [JIRA] (ASTERISK-21976) Set more than one codec in dialplan execution using SIP_CODEC (adapted chan_sip:try_suggested_codec)

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21976?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=207847#comment-207847 ] 

Matt Jordan commented on ASTERISK-21976:
----------------------------------------

While it's fine to have a patch on the issue for a released branch - which can be useful for users of Asterisk - since this is an improvement/new feature, it must be written against Asterisk trunk to be included in the next version. Please make sure your patch applies cleanly to trunk and attach it to the issue.

Some [coding guideline|https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines] problems:

# {{const char *codecList;}} => should be {{codec_list}}. Asterisk doesn't use camelCase or CamelCase.
# All {{if}} statements should have brackets. Older portions of the codebase didn't always follow this rule, but new code should.

Minor nitpick:

# Reduce indentation here:
{noformat}
+		ast_getformatbyname(ast_strip(codec), &fmt);
+		if (fmt.id) {
{noformat}
by using logging a NOTICE and {{continue}} if {{fmt.id}} is 0.

                
> Set more than one codec in dialplan execution using SIP_CODEC (adapted chan_sip:try_suggested_codec)
> ----------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-21976
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21976
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: SVN
>         Environment: FreeBSD (AMD64) 9.1 using Asterisk provided via ports (at the moment 11.4)
>            Reporter: Dennis Guse
>            Assignee: Dennis Guse
>            Severity: Minor
>         Attachments: patch-channels-chan__sip.c-393126
>
>
> For video calls, we would like to set the codecs in the dialplan using 
> SIP_CODEC. However, if SIP_CODEC is set, all codecs except the ONE set are disallowed and thus either audio or video is available.
> Attached is a patch for 11.4 that allows SIP_CODEC to contain a list of codecs , e.g. "gsm,h264".
> Patch against: 11.4: https://docs.google.com/file/d/0ByFooYVveHXdNng4WThoNV8zLVk/edit?usp=sharing
> Do you have any feedback?
> Authors: Dennis Guse (dennis.guse at qu.tu-berlin.de) und Frank Haase (fra.haase at googlemail.com).

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