Our provider has implemented RTP latching. It was causing us audio issues. This was a message from our inbound SIP vendor.
Our new platform required RTP latching, where you initiate the RTP stream, however that is not happening. Our engineering team rolled back this change for the time being. Can you please retest and keep us posted on your results?
I can’t find anything on this. Any information would help.
hmm I believe they are referring to rfc7362
sound like a variation of ICE, my best guess it to look at how you handle NAT on your setup
Configuring res_pjsip to work through NAT - Asterisk Project - Asterisk Project Wiki
can you get them to provide some documentation on how to implement it
or at least witch RFC you must support
my provider have me adding this to our calls
same => n,Set(PJSIP_HEADER(add,P-Early-Media)=supported) ; rfc5009
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