I've been trying to make srtp->srtp work with kamailio, rtpproxy and two symbian phones. rtpproxy does not receive any rtp/sdp packets. I'm still trying to debug what makes it fail. Without tls and srtp everything works, with either results are varied. Before I waste more time: Should kamailio+rtpproxy work out of the box with NATted TLS+SRTP clients? mediaproxy-ng is only needed for *conversion* between srtp and rtp?
On 7/19/13, Alexey Rybalko alexey.rybalko@gmail.com wrote:
Today I compiled and installed your branch from Kamailio repository. Another hope for kickstart was to swap Kamailio inside NGCP: in order to use already working config straight from then authors J But it was figured out that a lot of symbols and funcs are out of place (despite that all necessary modules are compiled). Then I found missing patches at Sipwise repo. But haven't been sure they are needed for the purpose or not.
p.s. Sorry if I'm annoying. Despite a hot set of projects for "WebRTC to SIP" convergence (almost) none of them are stable or working as expected. E.g. Asterisk doesn't (yet) officially support VP8 for B2BUA. And I don't want to involve Asterisk here as well.
regards, /A
2013/7/19 Richard Fuchs rfuchs@sipwise.com
On 07/19/13 14:46, Alexey Rybalko wrote:
Good! When NGCP 3.0 will be available for the community?
Have we any chance to evaluate media profile conversion(SDP) prior that event using base Kamailio? There a several patches from Sipwise for Kamailio core and some other modules as well (e.g. nathelper).The only examples for new mediaproxy might be found inside SPCE (NGCP), but it's config doesn't work without patching the Kamailio sources.
You don't have to patch it if you check out the rtpproxy-ng branch from the Kamailio git repo (or the Sipwise Kamailio repo for that matter), but you're still gonna have to compile the sources yourself. At this point, there's no way around that. NGCP 3.0 is on the horizon, but not quite there yet as there's still some issues that need to be worked out.
cheers
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