Hello, all.
I would know why we should avoid using the exec module exported functions outside out the request_route and on_failure_route blocks? (and so ... why it is forbidden).
Answers carry useful informations (e.g. SDP) that we could want to parse with an external command.
Maybe is there a more appropriate way to achieve this, and I mail you for this reason.
Thanks once again,
JF Smigielski
________________________________________________________________________
iBELGIQUE, exprimez-vous !
http://web.ibelgique.com/
Hi all,
I need a softphone that supports redirect. Let me explain:
In all free softphones I download, I have an "outbound proxy" field.
So, if my server returns and 302 Moved Temporarily, the softphone will
get the Contact header field and try to call the user again using the
outbound proxy, and the outbound will reply with an 302 and I will get
an loop.
I need a softphone who will use a location server to query the
location of a user and after that the softphone will make the call
directly, without using a proxy.
Could anyone point me a softphone that will do this?
Best Regards,
Felipe
--
Master Student - Electrical Engineering Department
Computer Engineering and Telecommunications Research Group
Universidade Federal de Minas Gerais - Brazil
"Come to me, all you who are weary and burdened, and I will give you rest."
Matthews 11:28
Hello, When I try to run the ser_ctl command to do something like
ser_ctl version or any other command, I get the error
ser_ctl: (111, 'Connection refused'): error
SER is running while I type the command.
Any hints will be appreciated.
_________________________________________________________________
Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with
Windows Media Player. Just Click PLAY.
http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006
Thank you. settig the xcap_root for the xcap module fixed the two errors
relating to xcap_root.
BUT I am still getting the two errors relating to the lookup_domain command,
namely:
-------------------------------------
ser: parse error (192,24-25): unknown command, missing loadmodule?
ser: parse error (261,26-27): unknown command, missing loadmodule?
ser: ERROR: bad config file (2 errors)
ser startup failed
----------------------------------
I am using the presence example file. The excerpt from my ser.cfg file is (I
have marked the LINES where the error occurs):
---- start ---
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/avp.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/dialog.so"
loadmodule "/usr/local/lib/ser/modules/rls.so"
loadmodule "/usr/local/lib/ser/modules/pa.so"
loadmodule "/usr/local/lib/ser/modules/xcap.so"
loadmodule "/usr/local/lib/ser/modules/presence_b2b.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/fifo.so"
loadmodule "/usr/local/lib/ser/modules/xmlrpc.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
{ LINES TRUNCATED}
if (uri==myself) {
if (!lookup_domain("To")) { <--------------------------
LINE 192
xlog("L_ERR", "Unknown domain to: %tu from: %fu\n");
route(1);
break;
}
if (method=="SUBSCRIBE") {
if (!t_newtran()) {
sl_reply_error();
break;
};
(lines truncated)
if (lookup_domain("From")) { <---------------------- LINE 261
if (lookup_user("From")) {
if
(is_simple_rls_target("$uid-list")) {
# log(1, "it
is simple subscription!\n");
# takes From
UID and makes XCAP query for user's
# list named
"default"
if
(!query_resource_list("default")) {
t_reply("404", "No such user list");
break;
}
}
}
}
----------- END ---------------------
Any suggestions??
THanks in advance
>From: samuel <samu60(a)gmail.com>
>To: "SER LIST" <sergrp(a)hotmail.com>
>CC: serusers(a)iptel.org, serdev(a)iptel.org
>Subject: Re: [Serusers] SER pre-release - PRESENCE ISSUES
>Date: Fri, 10 Nov 2006 10:14:17 +0100
>
>XCAP parameters where removed from rls and pa and a new module, called
>xcap contains the xcap root so you have to delete pa and rls xcap
>parameter, load the new module xcap.so and set the xcap_root parameter
>of xcap module to the directory having the doc structure.
>
>About the lookup_domain, there can be a typo. Can you post the config
>file syntax??
>
>Hope it helps!
>
>Samuel
>
>2006/11/10, SER LIST <sergrp(a)hotmail.com>:
>>I installed the SER pre-release along w/ all modules. The problem is that
>>presence now does not run (it was working w/ the presence-release-10.99).
>>The errors I get are:
>>----------------------------------------------------------------------------------------------------------
>>set_mod_param_regex: parameter <xcap_root> not found in module <rls>
>>ser: parse error (82,20-21): Can't set module parameter
>>set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa>
>>parse error (97,20-21): Can't set module parameter
>>ser: parse error (186,24-25): unknown command, missing loadmodule?
>>parse error (255,26-27): unknown command, missing loadmodule?
>>ERROR: bad config file (4 errors)
>>-------------------------------------------------------------------------------------------------------------
>>I checked the pa module code and also the rls code and the parameters
>>auth_xcap_root and xcap_root are NOT defined !!!!
>>
>>The error on line 185 and 255 refer to the "Lookup_domain" function. I
>>verified that the domain module has been compiled and included in the
>>ser.cfrg file
_________________________________________________________________
Ready for the world's first international mobile film festival celebrating
the creative potential of today's youth? Check out Mobile Jam Fest for your
a chance to WIN $10,000! www.mobilejamfest.com
I have my SER server running on a Linux machine and I am attempting to
forward some requests to a Cisco SIP server. I have been running X-lite
clients on windows and the same Linux box that the SER server is running on.
I have no problem completing calls between the X-lite clients OR with either
X-lite client calling the Cisco SIP server directly. However, when I try to
make a call from either X-lite client through the SER server I eventually
get timeout methods. I am trapping those calls and wrote the following
route block:
route [5] {
record_route();
forward(uri:host, uri:port);
}
I have tried replacing the forward with send() and t_relay() method calls,
but nothing seems to work.
The header from the X-lite client directly to the Cisco SIP server, which
works is:
INVITE sip:2158200@65.221.7.xxx SIP/2.0
Via: SIP/2.0/UDP
166.34.149.xxx:5061;rport;branch=z9hG4bK5963AD09C90F6F3A16F55A271C98F573
From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=1279481120
To: <sip:2158200@65.221.7.xxx>
Contact: <sip:1000@166.34.149.xxx:5061>
Call-ID: 28E5B723-9B02-BF38-1A81-1FECC0A85D5E(a)166.34.149.xxx
CSeq: 6254 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 313
The header from SER to the Cisco server, which does not work is:
INVITE sip:2158200@65.221.7.xxx SIP/2.0
Record-Route: <sip:166.34.149.xxx;ftag=2061592395;lr=on>
Via: SIP/2.0/UDP 166.34.149.xxx;branch=0
Via: SIP/2.0/UDP
166.34.149.xxx:5061;rport=5061;branch=z9hG4bK4EA9A90B8CE7CE52BFB69DA431560E5
7
From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=2061592395
To: <sip:2158200@65.221.7.xxx>
Contact: <sip:1000@166.34.149.xxx:5061>
Call-ID: 38CA166F-59A6-E44B-F34E-9E1D6307698B(a)166.34.149.xxx
CSeq: 3009 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 313
The only thing that I can see different is the record_route record and an
additional Via record.
Does anyone have any idea what I am doing wrong?
Thanks
Harry Harcrow
I am new to setting up SER and this is the test setup.
1. DHCP running from router
2. SJ Phone on laptop 192.168.X.X
3. Grandstream Budgetone 101 192.168.X.X
4. SIP Express Router 192.168.X.X
5. Asterisk Server 192.168.X.X attached to tdm400p card is an analog phone
Will this work internally? There is no connection to a telco.
The SJ phone is showing the following: nat/firewall: Port Restricted Cone
NAT
SIP calls use outbound proxy: 192.168.X.X (SER)
Will I have to register the SJ, Grandstream and Asterisk PBX?
Ed
Hi Users ,
I am Using SER-0.96 , When I make a call from
SER to Pstn it works fine I mean is all "INV"BYE"CAN".
But when I recieved a call from Pstn I cannot cancel the call because
X-lite is sending to pstn gateway 480 temp. service not available instead
of 486 Busy Here.
I am using default config. but I dont understand where
it went wrong .
Any Idea or help greatly appreciated
Thank You.
Regards,
Ravi.
Hi Kristian.
Thanks for your email.
I am using ser 0.9.6 and did not find uac_replace_from function. Then I
did a search and found that it is in openser. I did install openser and it
worked.
Do you or anyone know if this function will be added into ser release ?
Please advise, thanks,
Larry
>------------------------------
>Message: 20
>Date: Wed, 25 Oct 2006 13:11:06 +0200
>From: Kristian Larsson <kristian(a)netatonce.se>
>Subject: Re: [Serusers] Re: How to change Caller ID in FROM field ?
>To: "lawrence k.y. lin" <lawlin888(a)hotmail.com>
>Cc: serusers(a)lists.iptel.org
>Message-ID: <20061025111106.GJ30882(a)juniks.net>
>Content-Type: text/plain; charset=iso-8859-15
>
>On Tue, Oct 24, 2006 at 03:45:01PM -0700, lawrence k.y. lin wrote:
>>Hi all,
>>
>> Appreciate if someone can give me a suggestion or even an answer.
>> Since the call from SIP device to PSTN phone did not have Caller ID
>>and our Telco carriers request for the Caller ID, we have to set a number
>>before the calls are routed to PSTN gateway.
>> I did it in Asterisk before with the function calls SetCallerID(CLID)
>>or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these
>>function calls worked. But because of the scalabliity issue, I changed to
>>SER and would like to do the similar thing as in Asterisk.
>> I read the SER document and awared SIP_HF_FROM has the data for the
>>entired FROM field. Is there any other variables for Caller ID only ?
>> I searched the mailing list for SER and did not find any related topic
>>(only found one with title: caller-id with raius using sip-rpid, but it is
>>irrelevant).
>There is a function called uac_replace_from which
>can replace values in the From header, however I
>would need some help with it myself.
>For example, towards one of my SIP "peers", I need
>to prefix a "0" on all From headers which is no
>problem, just do a uac_replace_from("0$fU","sip:0$fU@$fd");
>$fU is the from user part and $fd is the from
>domain...
>But how do I do if I first want to strip a few
>characters of from $fU and then add a few others?
>Regards,
> Kristian.
>--
>Kristian Larsson
_________________________________________________________________
Find a local pizza place, music store, museum and more then map the best
route! http://local.live.com?FORM=MGA001
Hi list,
I am trying to realise dynamic invite-timeout-timers loading timer-settings
per call from my database using avp. It seems to work perfectly on my
development-system but fails nearly completly in my high loaded
life-environment. According to my logs, timer values are set correctly (eg.:
reply_received: FR_INV_TIMER = 15, see modules/tm/t_reply.c:1371), but most
of the times the timers have no effect.
My setup is similar to
http://siprouter.onsip.org/doc/gettingstarted/ch09.html, but avps are not
written by hand, but acquired with:
modparam("tm", "fr_inv_timer_avp", "mytimer");
avp_db_load("$ruri/username","s:mytimer");
'mytimer' is an attribute in an usr_preferences-table.
Ser's version is 0.9.6.
I already searched the list-archives without finding anything regarding this
issue. I Also failed accessing the bug-tracker (http://bugs.sip-router.org/),
it gave me an internal server error.
Has anybody experienced this effect? Is this a bug? Is there a workaround?
Best regards,
Marcus Hunger
I have a scenario where user Alice registered with OpenSER with UDP
port 5070. Then, I have user Bob sending an invite to Alice via the
OpenSER using TCP. Hence, OpenSER is bridging my TCP SIP requests
to UDP (and vice versa for the responses).
Here is the sequence:
Alice ----- TCP Invite -----> OpenSER ---------- UDP Invite ------
> Alice
Alice responds back with 180 ringing, then 200 Ok.
Upon getting the 200 OK, Bob sends an ACK (to complete the INVITE),
followed by an Immediate BYE.
THE TROUBLE IS...... the ACK and the BYE that Bob sends are dropped
by OpenSER, and never forwarded onto Alice!!!! I see nothing wrong
with the messages. WHat is happening?
sip:~ russdaigle$ sudo tcpdump -i en0 -A -q -s 2000 port 5060
Password:
tcpdump: verbose output suppressed, use -v or -vv for full protocol
decode
listening on en0, link-type EN10MB (Ethernet), capture size 2000 bytes
The following shows an sip-invite-bye from mutator using TCP to mjsip
(via
OPENSER). Everything after the ACK is dropped and not sent on.
Then, is a similar but UDP exchange. In that case, we get a 404 not
found
in response to the BYE!!!
01:17:59.553909 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 588
INVITE sip:alice@siptest2.com:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
To: Alice <sip:alice@siptest2.com:5060>
CSeq: 1 INVITE
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
Contact: <sip:bob@10.10.2.6>;transport=TCP
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 195
v=0
o=- 817933771 817933775 IN IP4 10.10.2.6
s=darkness
c=IN IP4 10.10.2.6
t=0 0
m=audio 5000 RTP/AVP 0 4 97
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000/1
a=rtpmap:97 telephone-event/8000
01:17:59.554929 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 536
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
To: Alice <sip:alice@siptest2.com:5060>
CSeq: 1 INVITE
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
Server: OpenSer (1.1.0-notls (i386/darwin))
Content-Length: 0
Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11962
req_src_ip=10.10.2.6 req_src_port=50135 in_uri=sip:alice@siptest2.com:
5060 out_uri=sip:alice@10.10.2.6:5070 via_cnt==1"
01:17:59.555035 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 814
INVITE sip:alice@10.10.2.6:5070 SIP/2.0
Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Record-Route: <sip:
10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
To: Alice <sip:alice@siptest2.com:5060>
CSeq: 1 INVITE
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
Contact: <sip:bob@10.10.2.6>;transport=TCP
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 195
P-hint: usrloc applied
v=0
o=- 817933771 817933775 IN IP4 10.10.2.6
s=darkness
c=IN IP4 10.10.2.6
t=0 0
m=audio 5000 RTP/AVP 0 4 97
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000/1
a=rtpmap:97 telephone-event/8000
01:17:59.667962 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 364
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
To: Alice <sip:alice@siptest2.com:5060>
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
CSeq: 1 INVITE
Server: mjsip stack 1.6
Content-Length: 0
01:17:59.699680 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 505
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Record-Route: <sip:
10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
To: Alice <sip:alice@siptest2.com:5060>
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
CSeq: 1 INVITE
Server: mjsip stack 1.6
Content-Length: 0
01:17:59.699683 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Record-Route: <sip:
10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
CSeq: 1 INVITE
Contact: <sip:alice@192.168.0.3:5070>
Server: mjsip stack 1.6
Content-Length: 135
Content-Type: application/sdp
v=0
o=- 817933771 817933775 IN IP4 10.10.2.6
s=darkness
c=IN IP4 192.168.0.3
t=0 0
m=audio 21000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
01:17:59.700273 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 440
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Record-Route: <sip:
10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
To: Alice <sip:alice@siptest2.com:5060>
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
CSeq: 1 INVITE
Server: mjsip stack 1.6
Content-Length: 0
01:17:59.700299 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 663
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Record-Route: <sip:
10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
CSeq: 1 INVITE
Contact: <sip:alice@192.168.0.3:5070>
Server: mjsip stack 1.6
Content-Length: 135
Content-Type: application/sdp
v=0
o=- 817933771 817933775 IN IP4 10.10.2.6
s=darkness
c=IN IP4 192.168.0.3
t=0 0
m=audio 21000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
01:17:59.804566 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435
ACK sip:alice@siptest2.com:5060 SIP/2.0
Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK59972
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183
CSeq: 1 ACK
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
Max-Forwards: 70
01:17:59.839493 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435
BYE sip:alice@siptest2.com:5060 SIP/2.0
Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK68444
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183
CSeq: 2 BYE
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
Max-Forwards: 70
01:18:00.199306 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1
Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043
Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah>
Record-Route: <sip:
10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah>
To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183
From: Bob <sip:bob@siptest2.com>;tag=fromhackblah
Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com
CSeq: 1 INVITE
Contact: <sip:alice@192.168.0.3:5070>
Server: mjsip stack 1.6
Content-Length: 135
Content-Type: application/sdp
v=0
o=- 817933771 817933775 IN IP4 10.10.2.6
s=darkness
c=IN IP4 192.168.0.3
t=0 0
m=audio 21000 RTP/AVP 0
a=rtpmap:0 PCMU/8000