Closed celevra closed 9 years ago
Looks like the formating of codecs is changed. ast_format_set is nolonger available dst_format and src_format are nolonger available
Same issue. Is this branch still maintained, or should we fork it for 13, since Asterisk 12 is not LTS and is bug-fix-only at this point?
Feel free to fork, I don't have the time currently to fix this for 13.
Oh i have same problem Vp8 passthrough on webrtc ..is not working actually media is going to the client but when Web Browser try to decode it raise and ERROR. the curious thing is that when you use a softphone like linphone VP8 is working fine seems like format used by asterisk doesn't liked ! same clients with patched asterisk 11 or 12 all work.
Opus support for Asterisk 13 has been implemented: https://github.com/seanbright/asterisk-opus/commit/de820fbc4a79cddf1dfd5a2167ea3c8c2db76b78
@goseeped vp8 passthrough should be implemented directly in Asterisk 12+. If you are having an issue related specifically to this repository, please open a new issue and clearly explain the problem (including affected versions).
@seanbright Thanks for the new patch i will tested , this should work for both channels ? i mean for chan_pjsip and chan_sip ? or it is only for chan_sip ? Thanks
I don't see why it wouldn't work with chan_pjsip
, but it definitely works with chan_sip
. If you have problems with chan_pjsip
open a new issue and describe the problem and I will take a look.
thank you for this nice patch, helped me a lot of times, but now we want to use asterisk 13. do you have the time to get the patch working again? i get these failures Hunk #1 FAILED at 1029. Hunk #2 FAILED at 1111. 2 out of 2 hunks FAILED -- saving rejects to file main/frame.c.rej
regards