On Thu, Aug 1, 2019, at 7:30 PM, James Cloos wrote: > I tried once again to switch an asterisk install to use pj instead of > chan_sip, and am seeing an odd error. > > This particular one is dealing with unwanted calls (endless sewer style > of reply), and given this sdp: > > v=0 > o=200107170215185:5060 16264 18299 IN IP4 0.0.0.0 > s=pplsip > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 25282 RTP/AVP 100 6 0 8 3 18 5 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > pj returns PJMEDIA_SDP_EMISSINGRTPMAP, which means it cannot find an > rtpmap line. Even though two exist. > > I see that example has 0.0.0.0 for the o and c lines, so obviously the > attacker doesn't want audio. But I'd still like to waste some of their > cycles dealing with a series of 180s. > > Chan_sip would send the call to the dialplan. > > Should pj not also?
It's not as tolerant/forgiving as chan_sip. It's following this from the RFC I believe: For each media format of that type supported by the agent, there SHOULD be a media format listed in the "m=" line. In the case of RTP, if dynamic payload types are used, an rtpmap attribute MUST be present to bind the type to a specific format. There is dynamic payload 100 in the SDP, but no rtpmap to state what it is thus violating that part of the SDP, and likely why the pjmedia-sdp code doesn't like it. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
