[asterisk-bugs] [JIRA] (ASTERISK-25160) [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Wed Jul 27 10:20:04 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25160?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team updated ASTERISK-25160:
-------------------------------------
Target Release Version/s: 14.0.0
> [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
> --------------------------------------------------------------------
>
> Key: ASTERISK-25160
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25160
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/CodecHandling
> Affects Versions: 13.6.0
> Reporter: Alexander Traud
> Severity: Minor
> Target Release: 13.7.0, 14.0.0
>
> Attachments: A_generate_sdp_fmtp_always.patch, B_register_interface_with_cached.patch
>
>
> Sometimes, Asterisk does not send a line {{x=fmtp:y}} while negotiating the formats via SIP/SDP. This affects all formats with an attribute module like H.263, H.264, and Opus {{res/res_format_attr_opus.c}}.
> This happens, because {{ast_format_generate_sdp_fmtp}} is called with a format without an attribute module (variable {{interface}} is NULL). This happens, because such a format was created before its attribute module got registered. This happens for
> I) cached formats: main » ast_codec_builtin_init » CODEC_REGISTER_AND_CACHE » ast_format_create | ast_format_cache_set » set_cached_format
> II) local capabilities: main » ast_local_init » ast_format_cap_append_by_type » ast_format_create
> Steps to reproduce:
> # {{make install samples}}
> # edit the configuration file {{sip.conf}} to allow {{opus}}
> # dial echo-test extension (600)
> *Question to the Asterisk team:*
> Why are format-attribute modules loaded so late, after channels are loaded?
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