On 23 Jun 2014, at 14:24, Joshua Colp <[email protected]> wrote: > Olle E. Johansson wrote: >> On 23 Jun 2014, at 13:59, SVN commits to the Digium >> repositories<[email protected]> wrote: >> >>> + /* If ICE negotiation is enabled the DTLS Handshake will be performed >>> upon completion of it */ >>> +#ifdef USE_PJPROJECT >>> + if (!rtp->ssl || rtp->ice) { >>> +#else >>> if (!rtp->ssl) { >>> +#endif >>> return 0; >>> } >>> >>> >> >> Does this ifdef mean I can not use chan_sip and chan_pjsip at the same time? >> If so, I think it's a bad solution. > > No. This branch is against 11 and res_rtp_asterisk has special logic so if > the dependencies for pjproject are not met then it will still be built, just > without ICE/STUN/TURN support. It used to be a hard requirement but due to > feedback from the community I changed it to be optional in a past life.
Aha, so it has nothing to do with the SIP channel at all. Ok then. Continue as if this conversation has not happened. /O :-) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
