I saw at https://github.com/64characters/Telephone/issues/453 and at https://stackoverflow.com/questions/62045516/how-to-change-rtp-media-stream-address-in-invite-sdp-using-pjsip
And I didn’ t understang where, which conf file in this option allow_sdp_nat_rewrite should be set up to TRUE.
Can somebody explain ?
This sounds like it is a PJSIP option not an Asterisk one. If I understand what it does (Google is returning pages that 403 for me, and not offering cached versions, so I only have their summary), the nearest Asterisk equivalent is symmetric RTP, however it is not an exact equivalent; it requires some incoming media and learns the address from that.
What the Google summaries suggest is that the option causes the Via address to be used, rather than than the c= address, but in most incorrectly compensated NAT situations, the Via address is wrong, as well.
Asterisk chan_pjsip uses PJPROJECT, but doesn’t expose all of it.
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