[sr-dev] Websocket testing

[Carlos Ruiz Díaz]

Hello,

I have a few questions about the websocket module which I have successfully
installed but I wasn't able to make calls through it:

1. After setting up the proxy ip:port in the call.htm file (of sipml5) to *
127.0.0.1:5060* the client started to work but kamailio script refused to
establish my connection because the following condition was not satisfied:


*if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT) {*
*                xlog("L_WARN", "HTTP request received on $Rp\n");*
*                xhttp_reply("403", "Forbidden", "", "");*
*                exit;*
*}*

*MY_WS_PORT* and *MY_WSS_PORT *are set to 80 and 443 respectively, as the
default config example of websocket module says so.

Then, I decided to change the ip:port to *127.0.0.1:80*, always in the
call.htm file and afterwards the condition was satisfied but sipml5 dies
with

 SIP stack start: proxy='127.0.0.1:80', realm='<sip:127.0.0.1>',
impi='2000', impu='<sip:2000 at 127.0.0.1>'
tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
 Connecting to 'ws://127.0.0.1:80'
tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
 Stack starting
tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
 Unexpected response code: 200 :1 <http://127.0.0.1/>
 __tsip_transport_ws_onerror
tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
 __tsip_transport_ws_onclose
tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5>
 Failed to connet to the server

Finally, I ended up commenting the condition block and restored the
original values of ip:port to *127.0.0.1:5060* .

Having done that, I tried again and another error was thrown but this time,
in the next condition block: * if ($hdr(Host) == $null ||
!is_myself($hdr(Host))) *

<script>: WebSocket
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <script>:  Host:
127.0.0.1:5060
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <script>:
 Origin: http://127.0.0.1
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [
127.0.0.1:5060] == [127.0.0.1]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [
127.0.0.1:5060] == [127.0.0.2]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [
127.0.0.1:5060] == [192.168.10.95]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [
127.0.0.1:5060] == [192.168.10.55]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [
127.0.0.1:5060] == [127.0.0.1]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [
127.0.0.1:5060] == [127.0.0.2]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [
127.0.0.1:5060] == [192.168.10.95]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [
127.0.0.1:5060] == [192.168.10.55]
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core>
[forward.c:462]: *check_self: host != me*
Aug  8 11:30:53 carlosrdcnx-laptop kamailio[16238]: WARNING: <script>: Bad
host 127.0.0.1:5060

I commented the block too and only then sipml5 was able to register itself.

What am I doing wrong here?

2. I registered a legacy softphone (twinkle) to attempt to initiate a call
in both ways, but the was something wrong with the signaling, probably some
frame decoding garbage in the buffer of the SIP message. Perhaps these
bytes are part of the frame control header but since I haven't read the RFC
(yet) I am mentioning it anyway.

tcp_send: buf=*#012�~#003�*INVITE sip:2000 at df7jal23ls0d.invalid;transport=ws
SIP/2.0#015#012Record-Route:
<sip:127.0.0.1;transport=ws;r2=on;lr=on>#015#012Record-Route:
<sip:127.0.0.1;r2=on;lr=on>#015#012Via: SIP/2.0/WS
 127.0.0.1;branch=z9hG4bK90a8.b1a7035e13ed19880dd12a1f4c86adbb.0#015#012Via:
SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=z9hG4bKimixlbyp#015#012Max-Forwards:
69#015#012To: <sip:2000 at 127.0.0.1>#015#012From: "1000"
<sip:1000 at 127.0.0.1>;tag=lrtfz#015#012Call-ID:
gxsqobolphfchfq at carlosrdcnx-laptop.site#015#012CSeq: 654
INVITE#015#012Contact: <sip:1000 at 127.0.0.1:5062>#015#012Content-Type:
application/sdp#015#012Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE#015#012Supported:
replaces,norefersub,100rel#015#012User-Agent:
Twinkle/1.4.2#015#012Content-Length:
302#015#012#015#012v=0#015#012o=twinkle 391470222 1383232165 IN IP4
127.0.0.1#015#012s=-#015#012c=IN IP4 127.0.0.1#015#012t=0 0#015#012m=audio
8008 RTP/AVP 98 97 8 0 3 101#015#012a=rtpmap:98
speex/16000#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:3
GSM/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=ptime:20#015#012

3. Does Twinkle support the minimum media requirements for testing? If not,
what (Linux) softphone is suitable for this purpose?

Thanks and sorry for the long email.

Carlos.
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