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seanbright / asterisk-opus

Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
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Opus works but WARNINGS detected #5

Open ovoshlook opened 9 years ago

ovoshlook commented 9 years ago

I successfully installed patch and checks it work. IT works fine but asterisk cli sends this messages when rtp sended. [2015-02-05 00:53:01] WARNING[58036][C-00000005]: codec.c:363 ast_codec_samples_count: Unable to calculate samples for codec opus [2015-02-05 00:53:01] WARNING[58036][C-00000005]: translate.c:392 framein: no samples for opustolin48

ghost commented 9 years ago

I saw that first error using passthrough (no opus codec installed on asterisk) and chan_sip. Asterisk v13.2.0. Just like you it was working but that first error kept repeating over and over several times a second from asterisk cli.

I did not see that error when running on an Asterisk system with opus codec installed and not using passthrough.

traud commented 9 years ago

shadowym, because you face that issue with pass-through already, did you open an issue directly with the Asterisk project? If not, please, create a new issue there, so the whole Asterisk community is aware of this. Furthermore, I am not able to reproduce your issue, yet. Therefore, please, write two or one sentences about your setup. For example which devices are connected via chan_sip and whether one is using encrypted TLS/SRTP or just plain UDP/RTP.

ovoshlook, if you have the opportunity, remove codecs/codec_opus.c, recompile, and test pass-through only. If the same happens, please, do not wait for shadowym but create an issue with the Asterisk project as soon as possible. And do not forget to mention just your issue ID here, so the issues are linked and tracked.

traud commented 8 years ago

Do you still face that issue, even in Asterisk 13.12 with opus_codec module from Digium?