Hello, I am trying to conference an existing RTP audio stream with a call made via pjsua2. I've looked through the mailing list and others have suggested using the streamutil sample as a starting point. I have tried adapting that code inside a new AudioMedia subclass, however the create_stream() crashes at pjmedia_stream_create() with a floating point exception because of a division by zero when it calls an underlying rtp/rtcp method to calculate the NTP timestamp. Even removing the pjsua2 SIP call entirely and just trying to create the RTP stream alone right after the library is initialized still results in the same error. I've verified that the codec is set properly (PCMU) and the codec info has the proper clock rate set (8KHz), etc. One difference in create_stream() is that I use pjmedia_transport_udp_create2() instead of pjmedia_transport_udp_create() so that I can set the correct IP address, but the status for that returns successful. I am not sure if maybe it has to do with the fact that I'm mixing the pjsua2 API *and* using the low-level API at the same time? I should note I'm also using Endpoint::instance().audDevManager().setNullDev(); during initialization if that makes any difference. Is there an example anywhere of creating a non-file recorder and non-file player stream like this? Or does anyone have any suggestions as to what I should check as far as settings anywhere or something? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150307/3c4e5f1f/attachment.html>
