[asterisk-bugs] [JIRA] (ASTERISK-27008) H264 SDP parse fails if fmtp optional parameters have a space

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27008?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237044#comment-237044 ] 

Asterisk Team commented on ASTERISK-27008:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> H264 SDP parse fails if fmtp optional parameters have a space
> -------------------------------------------------------------
>
>                 Key: ASTERISK-27008
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27008
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_format_attr_h264
>    Affects Versions: 14.4.0
>            Reporter: John Harris
>
> Discovered while diagnosing a failure of Grandstream GXV3275 video phones to negotiate 720P resolution when SIP call goes via Asterisk.
> Also reported to Grandstream- https://forums.grandstream.com/forums/index.php?topic=35745
> The initial phone invite contains:
> a=fmtp:99 profile-level-id=42801F; packetization-mode=1
> Which Asterisk then reinvites the target phone as:
> a=fmtp:99 profile-level-id=42801F
> So the target phone replies:
> a=fmtp:99 profile-level-id=428014
> So profile 42801F was requested (H264 level 3.1, 1280x720), but callee responded with profile 428014 (H264 level 2.0, 352x288). These phones require that Asterisk preserves the packetization-mode when it sends the invite.
> Asterisk does actually try to parse the initial SDP contents (h264_parse_sdp_fmtp, res/res_format_attr_h264.c). This fails because it splits on semi-colon only, but the Grandstream firmware is also writing a space.
> RFC6184 (https://tools.ietf.org/html/rfc6184#page-57) states that these parameters should be 'in the form of a semicolon-separated list of parameter=value pairs'. However the examples in the RFC do also include spaces.
> As a workaround you can modify h264_parse_sdp_fmtp to split on semicolon and space:
> while ((attrib = strsep(&attribs, "; "))) {
> .. but clearly this will regress clients who do not use a space.



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