Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-sendto / 360_non_sip_uri.py
1 # $Id$
2 import inc_sip as sip
3 import inc_sdp as sdp
4
5 # Some non-SIP URI's in Contact header
6 #
7 complete_msg = \
8 """INVITE sip:localhost SIP/2.0
9 Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj9db9
10 Max-Forwards: 70
11 From: <sip:192.168.0.14>;tag=08cd5bfc2d8a4fddb1f5e59c6961d298
12 To: <sip:localhost>
13 Call-ID: 3373d9eb32aa458db7e69c7ea51e0bd7
14 CSeq: 0 INVITE
15 Contact: mailto:dontspam@pjsip.org
16 Contact: <mailto:dontspam@pjsip.org>
17 Contact: http://www.pjsip.org/the%20path.cgi?pname=pvalue
18 Contact: <sip:localhost>
19 User-Agent: PJSUA v0.9.0-trunk/win32
20 Content-Length: 0
21 """
22
23
24 sendto_cfg = sip.SendtoCfg( "Non SIP URI in Contact", 
25                             "--null-audio --auto-answer 200", 
26                             "", 200, complete_msg=complete_msg)
27