On Sun, 2012-10-07 at 00:29 +0100, Josh wrote: > > 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? Yes. > > 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? > I wasn't aware of this limitation. I haven't myself tested multiple accounts with different connection types. > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
