> The incoming INVITE offers only RTP, and you have configured mandatory > SRTP. So the call will not be successful. > Have you tried to configure optional SRTP, or only RTP? > Thanks Dan. It was 'Mandatory' as I wasn't fully aware of how the (S)RTP works and the fact that data streams opened between my UA and the person calling have nothing whatsoever to do with my SIP server. My SIP server does offer TLS/SRTP, hence why I thought, wrongly, that I could enforce that option. I will try the 'optional' setting and see if that works. On a side note, I presume that my TLS (i.e. the signalling between my UA and the SIP server) can still be enforced even if SRTP isn't, correct? One last query, if I may - my UA allows me to use multiple accounts at the same time. I haven't upgraded this for a while and at the time I was told that there is a pjsip limitation of having a mixture of TLS and non-TLS accounts (one needs to have one or the other, but not a mixture of both at the same time - in other words, I can't have TLS and non-TLS accounts in use at the same time). Has this limitation been addressed with newer versions of pjsip or is this still in place?
