[asterisk-bugs] [JIRA] (ASTERISK-25160) [patch]SIP: SDP: Audio-Codecs: Opus: rtpmap missing

[Alexander Traud (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25160?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Alexander Traud updated ASTERISK-25160:
---------------------------------------

    Attachment:     (was: B_register_interface_with_cached.patch)

> [patch]SIP: SDP: Audio-Codecs: Opus: rtpmap missing
> ---------------------------------------------------
>
>                 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: SVN, 13.4.0
>         Environment: Ubuntu 14.04 LTS
>            Reporter: Alexander Traud
>            Severity: Minor
>         Attachments: A_generate_sdp_fmtp_always.patch, B_register_interface_with_cached.patch
>
>
> Sometimes, Asterisk does not send a "rtpmap" line in 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 sip.conf to allow "opus"
> # dial echo-test extension (600)



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