i have yet to finish my readings on the websocket standards but just wanted to fire away with this question.
is the behavior of protocol conversions between UDP and TCP the same as if you include websockets?
Record-Route: <sip:192.168.122.100;r2=on;lr=on>
Record-Route: <sip:192.168.122.100:8080;transport=ws;r2=on;lr=on>
Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0
Via: SIP/2.0/WS z173czhz21tk.invalid;rport=54765;received=192.168.122.1;branch=z9hG4bK3818745
Max-Forwards: 16
Call-ID: 69hbgnng64at9p07r2j4
CSeq: 9406 INVITE
Contact: <sip:kelvin2@z173czhz21tk.invalid;alias=192.168.122.1~54765~5;transport=ws;ob>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.2.1
Content-Length: 2103
v=0
o=- 3148117784 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
m=audio 55736 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 192.168.122.1
a=rtcp:55736 IN IP4 192.168.122.1
a=candidate:2625852906 1 udp 2113937151
<cut off>
---
+++ 16-3-2013 10:41:25.584732 INFO SIP ::send_sip_udp
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745
Call-ID: 69hbgnng64at9p07r2j4
CSeq: 9406 INVITE
Server: Twinkle/1.4.2
Content-Length: 0
---
+++ 16-3-2013 10:41:25.589231 INFO SIP ::send_sip_udp
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745
Call-ID: 69hbgnng64at9p07r2j4
CSeq: 9406 INVITE
Server: Twinkle/1.4.2
Warning: 302 X340precise "Incompatible transport protocol"
Content-Length: 0