[asterisk-bugs] [JIRA] (ASTERISK-25579) Outgoing Packetization Time (Speex, AMR, Opus, …)

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25579?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228389#comment-228389 ] 

Asterisk Team commented on ASTERISK-25579:
------------------------------------------

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].

> Outgoing Packetization Time (Speex, AMR, Opus, …)
> -------------------------------------------------
>
>                 Key: ASTERISK-25579
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25579
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_speex, Codecs/General
>    Affects Versions: 11.20.0, 13.6.0
>            Reporter: Alexander Traud
>            Severity: Minor
>
> Since Asterisk 13, an ast_translator {{codecs/codec_*}} does not know the amount of frames to be put into one RTP payload packet in its routine frameout. However, this is required when the packetization time (in SIP/SDP: ptime) is not the default one.
> For example, when a ptime of 60ms is negotiated for Speex, Asterisk should send 60ms in one RTP packet. Currently, Asterisk sends three times 20ms.
> This affects all (only) those formats which cannot be smoothed:
> * Asterisk 11, see res/res_rtp_asterisk.c:ast_rtp_write
> * Asterisk 13, see main/codec_builtin.c; all codecs without ".smooth = 1"
> With those formats, ast_format_can_be_smoothed(.) returns 0, because the format uses
> (A) variable bit-rate (vbr) or
> (B) several modes with different bit-rates.
> In those cases, {{ast_format_get_minimum_bytes(.)}} is of no help to create the amount of frames per payload. These formats are Speex, G.719, [G.723.1|http://asterisk.hosting.lv/], [Siren|http://www.digium.com/products/asterisk/downloads], [SILK|https://github.com/traud/asterisk-silk], and CELT. Furthermore, this affects external modules like [iLBC 20|https://github.com/traud/asterisk-ilbc], [AMR(-WB)|https://github.com/traud/asterisk-amr], [Codec 2|https://github.com/traud/asterisk-codec2], and [Opus|https://github.com/seanbright/asterisk-opus].
> This is just about outgoing packets (Asterisk to another party). Incoming packets must be decodable by a transcoding module even without this information, because the incoming ptime could change at any time.
> {{ast_rtp_codecs_get_framing(.)}} provides this information. Furthermore, the channel driver negotiates this value, therefore the channel knows this as well. However, this information is required in {{ast_translator.frameout(ast_trans_pvt)}}.
> In Asterisk 11, this information is available. However, its Speex module does not us this to create the correct frame length, either.



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