[Core, mod_sofia] Make the rtp-autofix-timing codec check limit configurable - #3106
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RicardoMDomingues wants to merge 1 commit into
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[Core, mod_sofia] Make the rtp-autofix-timing codec check limit configurable#3106RicardoMDomingues wants to merge 1 commit into
RicardoMDomingues wants to merge 1 commit into
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…gurable The CBR branch of the autofix-timing codec check stopped after a hardcoded 50 frames. Add a sofia profile parameter, rtp-autofix-max-codec-check-frames: > 0 inspect that many frames (default 50, unchanged) < 0 inspect for the whole call
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Description
The CBR branch of the autofix-timing codec check stopped after a hardcoded 50 frames.
Add a sofia profile parameter, rtp-autofix-max-codec-check-frames:
Type of Change
Testing
Checklist
Additional Notes
I need this change because I use FreeSWITCH (FS) with Trader turret systems. In some setupus, media is routed to a media server, and the packet rate can change dynamically between all participants (from the Turret phones, to their gateway, through the customer's SBC, and finally to our system).
When the Turret system changes the packet rate, it does not send a SIP re-INVITE or any other SIP signaling to notify the gateway or SBC. Because of this, there is no way to detect the rate change via SIP. Instead, it can only be detected and handled using the RTP timestamp and payload information.
FreeSWITCH already had a feature to handle this, but it previously only worked at the very beginning of the call. My change extends this functionality so FreeSWITCH by making configurable how many packets we want to check and, if its negative, check for entire call