Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-sendto / 260_multipart_err_no_sdp.py
1 # $Id$
2 import inc_sip as sip
3 import inc_sdp as sdp
4
5 body = \
6 """
7 --12345
8 Content-Type: application/notsdp
9
10 v=0
11 o=- 0 0 IN IP4 127.0.0.1
12 s=pjmedia
13 c=IN IP4 127.0.0.1
14 t=0 0
15 m=audio 4000 RTP/AVP 0 101
16 a=rtpmap:0 PCMU/8000
17 a=sendrecv
18 a=rtpmap:101 telephone-event/8000
19 a=fmtp:101 0-15
20
21 --12345
22 Content-Type: text/plain
23
24 Hi there this is definitely not SDP
25
26 --12345--
27 """
28
29 args = "--null-audio --auto-answer 200 --max-calls 1"
30 extra_headers = "Content-Type: multipart/mixed; boundary=12345"
31 include = []
32 exclude = []
33
34 sendto_cfg = sip.SendtoCfg( "Multipart/mixed body without SDP", 
35                             pjsua_args=args, sdp="", resp_code=400, 
36                             extra_headers=extra_headers, body=body,
37                             resp_inc=include, resp_exc=exclude)
38