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DirectRTP Issues #417

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DirectRTP Issues #417

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@dkgroot dkgroot commented Oct 19, 2017

Fixes #391
Fixes #394
Fixes #405

Has to be tested by the issue owners

Only implemented in ast113 at the moment. Changes would have to be duplicated to other asterisk versions before committing the pull request.

- Split resume up into two parts (second part will be called by receiveChannelOpen
- When directrtp is active, update_rtp_peer should call startMediaTransmission (after receiveChannelOpen has completed)
- Call openReceiveChannel when RINGOUT is indicated (ignoring earlyrtp)
- Update indirect rtp handling according to the new method

References #391
References #394
References #405
Enh: integrate audio and video handling in one get_rtp_peer function
Enh: update_rtp_peer: check if ast_rtp_instance_get_remote_address / ast_rtp_instance_get_local_address return is not null
Fix: debug output (rtp_createServer)
Enh: Add location to call 'sccp_rtp_requestRTPPorts(device, channel);' in sccp_channel_newcall.
Fix: Tone indication is stopped by sccp_channel_receiveChannelOpen. So does not have to be repeated

References #391
References #394
References #405
References #394

Signed-off-by: Diederik de Groot <[email protected]>
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directrtp=on - no audio hold/resume call using directrtp Voice loopback when directrtp=on
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