[asterisk-bugs] [JIRA] (ASTERISK-9979) [patch] PCMA/16000 and PCMU/16000 support (hd telephony)

[Alexander Traud (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-9979?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236254#comment-236254 ] 

Alexander Traud commented on ASTERISK-9979:
-------------------------------------------

Short answer: I implemented this audio format at http://github.com/traud/asterisk-alaw16

Long answer: A bit late, but for other reasons, I had to implement what was discussed here, now ten years ago. This report was about ignoring the sample rate in SIP/SDP and therefore using PCMU/8000 instead of PCMU/xxxxx. Furthermore, when this report was created, several wide-band audio formats competed. The original reporter was employed at AVM Germany who create the Internet access devices (IAD) FRITZ!Box. Furthermore, at that time, AVM still used Wi-Fi instead of DECT to attach phones and therefore created a product called FRITZ!Mini – a Wi-Fi enabled SIP phone. That device did not use G.722 but PCMU/16000 for HD voice. To avoid transcoding, jitter, and echo, AVM used that PCMU/16000 even between their FRITZ!Box. Some SIP providers simply forward the SDP and therefore, the endpoints are able to negotiate directly. Because G.722 won the race, I am not adding this format to the core of Asterisk but provide it as external module. For those, who stumble across this report via an Internet search, please, have a look at the above hyperlink. It contains even further background information.

> [patch] PCMA/16000 and PCMU/16000 support (hd telephony)
> --------------------------------------------------------
>
>                 Key: ASTERISK-9979
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-9979
>             Project: Asterisk
>          Issue Type: New Feature
>          Components: Channels/NewFeature
>            Reporter: Guido R.
>         Attachments: hd_telephony.diff, pcma_16000.eth
>
>
> With this patch PCMA/16000 and PCMU/16000 will be accepted during the SDP negotiation. Prior asterisk ignored the sample rate in the rtpmap lines of the SDP body and changed the 16000 to 8000.. and so the negotiation between the UAs failed.
> Now the sample rate is parsed too, and for now PCMA/16000 and PCMU/16000 are supported by the new option flag AST_OPT_16000.
> ****** ADDITIONAL INFORMATION ******
> The data structure rtpPayloadType has been extended by a sample_rate attribute. The list of available codecs is extended by the sample rate attribute. For now only G722 has a sample rate of 16000.
> A new codec implemented in asterisk for the PCMA/16000 and PCMU/16000 is not needed.
> An asterisk test server is running in our company with this new feature enabled. Our tests with hd telephony equipment work fine.



--
This message was sent by Atlassian JIRA
(v6.2#6252)