[asterisk-dev] Odd PJMEDIA_SDP_EMISSINGRTPMAP

[James Cloos]

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?

-JimC
-- 
James Cloos <cloos at jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6