thank dave ,
as i said you earlies iam using the xlite
which is a stun client and iam running
the stun 0.92 + ser in same box
so in debug mode of stun i could able to see public ip
allocate to that private ip but ser is not showing
any difference ( this is also in the debug mode)
so i donot know where is the error
actualy to test my stun server i have installed
winstun in the private ip and when i use runtest
the message it gives is
Port restricted NAT detected - VoIP will work with
STUN Preserves port number Does not supports hairpin
of media Public IP address: 202.53.76.51
so from here how to move further even rtpproxy is not
working fine with this
my ser.cfg file is
*******************************************************
# This default script includes nathelper support. To
make it work
# you will also have to install Maxim's RTP proxy. The
proxy is enforced
# if one of the parties is behind a NAT.
#
# If you have an endpoing in the public internet which
is known to
# support symmetric RTP (Cisco PSTN gateway or
voicemail, for example),
# then you don't have to force RTP proxy. If you don't
want to enforce
# RTP proxy for some destinations than simply use
t_relay() instead of
# route(1)
#
# Sections marked with !! Nathelper contain
modifications for nathelper
#
# NOTE !! This config is EXPERIMENTAL !
#
# ----------- global configuration parameters
------------------------
debug=8 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.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/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
#loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which
true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping
interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping
only clients behind NAT
# ------------------------- request routing logic
-------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# !! Nathelper
# Special handling for NATed clients; first, NAT test
is
# executed: it looks for via!=received and RFC1918
addresses
# in Contact (may fail if line-folding is used);
also,
# the received test should, if completed, should
check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == "REGISTER" || !
search("^Record-Route:")) {
log("LOG: Someone trying to register from
private IP, rewriting\n");
# This will work only for user agents that
support symmetric
# communication. We tested quite many of them
and majority is
# smart enough to be symmetric. In some phones
it takes a configuration
# option. With Cisco 7960, it is called
NAT_Enable=Yes, with kphone it is
# called "symmetric media" and "symmetric
signalling".
fix_nated_contact(); # Rewrite contact with
source IP of signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active
to SDP
};
force_rport(); # Add rport parameter to topmost
Via
setflag(6); # Mark as NATed
};
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy;
that's
# particularly good if upstream and downstream
entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take
the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following
command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest
authentication
# if (!www_authorize("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our
USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if
(uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&& !search("^Route:")){
sl_send_reply("479", "We don't forward to private
IP addresses");
break;
};
# if client or server know to be behind a NAT, enable
relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all
transactions (for example,
# re-INVITEs from public to private UA are hard to
identify as
# NATed at the moment of request processing); look at
replies
t_on_reply("1");
# send it out now; use stateful forwarding as it
works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]")
{
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and
we did not
# know at time of request processing ? (RFC1918
contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
*******************************************************
with regards
serdiehard
--- Dave Bath <dave(a)fuuz.com> wrote:
> You should only need to enable stun from your client
> (obviously only
> works for a client which supports stun).
>
> Sorry, I do not use cpled
>
> -----Original Message-----
> From: ser die [mailto:serdiehard@yahoo.com]
> Sent: 16 August 2004 14:43
> To: Dave Bath
> Cc: serusers(a)lists.iptel.org
> Subject: RE: [Serusers] ser + stun
>
> yes dave,
>
> my stun server is working correctly i have tested
>
> with winstunsetup.msi
>
> now i need to work this with ser
>
> does i need to have any configurarion changes in the
>
>
> ser.cfg?
>
> has the cpled worked for you?
>
> with regards
> serdiehard
>
> --- Dave Bath <dave(a)fuuz.com> wrote:
>
> > Hey,
> >
> > Haven't really used the raw fifo commands, so
> can't
> > help you there. I
> > suggest you find a machine from which you can
> > confirm whether your stun
> > server is operating correctly before you worry
> about
> > the ser.cfg.
> >
> > Regards,
> >
> > Dave
> >
> > -----Original Message-----
> > From: ser die [mailto:serdiehard@yahoo.com]
> > Sent: 16 August 2004 13:52
> > To: Dave Bath
> > Subject: RE: [Serusers] ser + stun
> >
> > thanks dave,
> >
> > i have downloaded winstunsetup but the format is
> > .msi
> > mines is a linux system so i donot know how to use
> > is it and how to get the confirmation that stun is
>
> > correctly setup in the system
> >
> > one other thing do you have any idea about how to
> > run
> > the fifo commands?
> >
> >
> > with regards
> > serdiehard
> >
> > --- Dave Bath <dave(a)fuuz.com> wrote:
> >
> > > Have you used the WinStun program to confirm
> that
> > > your stun server is
> > > correctly set up?
> > >
> > > Dave
> > >
> > > -----Original Message-----
> > > From: serusers-bounces(a)lists.iptel.org
> > > [mailto:serusers-bounces@lists.iptel.org] On
> > > Behalf Of ser die
> > > Sent: 16 August 2004 08:44
> > > To: serusers(a)lists.iptel.org
> > > Subject: [Serusers] ser + stun
> > >
> > > hello friends,
> > >
> > > iam using ser of latest cvs head
> > >
> > > and stun server of 0.92 version
> > >
> > > iam running ser + stun on the same box
> > >
> > > mines problem is stun could able to communicate
> > >
> > > with the behind nat boxes and give an public ip
> to
> >
> > >
> > > that machines but ser could not able to even
> > >
> > > detect that.
> > >
> > > is there any special cofigurations we need to
> make
> > >
> > > with regards
> > > rama kanth
> > >
> > >
> > >
> > >
> __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam? Yahoo! Mail has the best spam
> > > protection around
> > > http://mail.yahoo.com
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> >
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > Take Yahoo! Mail with you! Get it on your mobile
> > phone.
> > http://mobile.yahoo.com/maildemo
> >
> >
> >
> >
> >
>
>
>
>
> __________________________________
> Do you Yahoo!?
> New and Improved Yahoo! Mail - Send 10MB messages!
> http://promotions.yahoo.com/new_mail
>
>
>
>
>
__________________________________
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Hello all,
Additional infor to below is I could run the sipsak
successfully. but just no audio could pass through the
NAT.
[root@detone stund]# sipsak -T -s
sip:1008@202.129.171.223
warning: IP extract from warning activated to be more
informational
0: 10.38.38.14 (3.749 ms) SIP/2.0 483 Too Many Hops
1: 219.95.43.92 "detected NAT type is full cone"
Contact (102.951 ms) SIP/2.0 200 OK
Contact:
<sip:1008@219.95.43.92:5060;user=phone>
[root@detone stund]#
--- "C.K" <ckng128(a)yahoo.com> wrote:
> Date: Sun, 15 Aug 2004 21:51:44 -0700 (PDT)
> From: "C.K" <ckng128(a)yahoo.com>
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Asterisk inside a NAT, client
> inside ANOTHER NAT
>
> Hello,
>
> By looking at this section from the link
>
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
>
> 9. Asterisk inside a NAT, client inside ANOTHER NAT
>
> In this case, we need a middle man to even find each
> other, an outbound SIP proxy that handles the SIP
> transaction and is reachable by all parties. To get
> media streams from point to point we need another
> middle man, a media server. Asterisk could be that
> media server, that could add media codec conversion.
> Portaone's rtpproxy works together with SIP Express
> router as a media server in this situation.
>
> Could anyone share the configuration on how to do
> this
> ? I could only succeed if I put on port forwarding
> on
> the UA's end.
>
> Regards, C.K
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Hi there...I got it working - all kinds of settings on the client side,
internal ip etc.
However quality is poor....
I assume some iptables settings on our SER machine and on the cisco
should be giving priority to voice over other data, is there a resource
on this?
karunb(a)omnitechnology.net wrote:
>Hi Dimitri,
> Did you got it working it is the NAT Issue with the SIP Proxy
>ALG which may be causing the Port disable after a while due to
>Dynimic Port allocation on the source side.
>
>If you still had the problem let me know.
>
>Regards,
>Karun
>
>
>
>
>>Hi everybody,
>>
>>I'm new to this list and currently trying to get things working with
>>ser. We have a ser server behind a cisco soho router (DSL) and I
>>enabled udp and tcp on ports 5060, but clients outside cannot register
>>with the server.
>>
>>When I dialin with a vpn connection, registration is no problem.
>>
>>Did anyone experience this problem? Isn't opening up 5060 enough or is
>>the problem in host name resolution. I'm seeing some syslog entries with
>>ser complaining that it cannot resolve 'dg' which isn't a hostname, but
>>my username.
>>
>>Thanks in advance for any help.
>>
>>Dimitri Georganas
>>MITC
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
yes dave,
my stun server is working correctly i have tested
with winstunsetup.msi
now i need to work this with ser
does i need to have any configurarion changes in the
ser.cfg?
has the cpled worked for you?
with regards
serdiehard
--- Dave Bath <dave(a)fuuz.com> wrote:
> Hey,
>
> Haven't really used the raw fifo commands, so can't
> help you there. I
> suggest you find a machine from which you can
> confirm whether your stun
> server is operating correctly before you worry about
> the ser.cfg.
>
> Regards,
>
> Dave
>
> -----Original Message-----
> From: ser die [mailto:serdiehard@yahoo.com]
> Sent: 16 August 2004 13:52
> To: Dave Bath
> Subject: RE: [Serusers] ser + stun
>
> thanks dave,
>
> i have downloaded winstunsetup but the format is
> .msi
> mines is a linux system so i donot know how to use
> is it and how to get the confirmation that stun is
> correctly setup in the system
>
> one other thing do you have any idea about how to
> run
> the fifo commands?
>
>
> with regards
> serdiehard
>
> --- Dave Bath <dave(a)fuuz.com> wrote:
>
> > Have you used the WinStun program to confirm that
> > your stun server is
> > correctly set up?
> >
> > Dave
> >
> > -----Original Message-----
> > From: serusers-bounces(a)lists.iptel.org
> > [mailto:serusers-bounces@lists.iptel.org] On
> > Behalf Of ser die
> > Sent: 16 August 2004 08:44
> > To: serusers(a)lists.iptel.org
> > Subject: [Serusers] ser + stun
> >
> > hello friends,
> >
> > iam using ser of latest cvs head
> >
> > and stun server of 0.92 version
> >
> > iam running ser + stun on the same box
> >
> > mines problem is stun could able to communicate
> >
> > with the behind nat boxes and give an public ip to
>
> >
> > that machines but ser could not able to even
> >
> > detect that.
> >
> > is there any special cofigurations we need to make
> >
> > with regards
> > rama kanth
> >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam? Yahoo! Mail has the best spam
> > protection around
> > http://mail.yahoo.com
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
>
> __________________________________
> Do you Yahoo!?
> Take Yahoo! Mail with you! Get it on your mobile
> phone.
> http://mobile.yahoo.com/maildemo
>
>
>
>
>
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Now to provide a complete answer to the question: the values are supposed to
be unique according to RFC3261 but there is nothing in the world that
prevents broken implementations to generate invalid non-unique values.
There are indeed implementations that are broken. To make CDRs as much
unique as you can, compose the identifier as a triple callid-fromtag-
totag. Composition of values generated by UACs and UASs significantly
reduces the chance of not being unique.
-jiri
At 06:23 PM 8/12/2004, Michael Shuler wrote:
>The call-id is supposed to be unique across a complete session i.e.
>INVITE/RINGING/OK/ACK/BYE. If it wasn't then there would be no way to
>relate the messages as part of the same sequence/call.
>
>----------------------------------------
>
>Michael Shuler, C.E.O.
>BitWise Systems, Inc.
>682 High Point Lane
>East Peoria, IL 61611
>Office: (217) 585-0357
>Cell: (309) 657-6365
>Fax: (309) 213-3500
>E-Mail: mike(a)bwsys.net
>Customer Service: (877) 976-0711
>
>> -----Original Message-----
>> From: serusers-bounces(a)lists.iptel.org
>> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Arnd Vehling
>> Sent: Thursday, August 12, 2004 10:47 AM
>> To: serusers(a)lists.iptel.org
>> Subject: [Serusers] From-Tag/Call-id not uniq?
>>
>>
>> Hi,
>>
>> according to the RFC:
>>
>> "When a tag is generated by a UA for insertion into a request or
>> response, it MUST be globally unique and cryptographically random
>> with at least 32 bits of randomness."
>>
>> But when looking at a sip logfile:
>>
>> egrep "^From:.*;tag=9d49295bb825210f" /var/log/term.log | wc -l
>> 503
>>
>> Question: Did i understood something wrong or is the recoccurance of
>> the same tag-id and callid in different sessions voilating the rfc?
>>
>> Ive found loads of non-uniq call-ids as well as non uniq tags. How
>> is a man supposed to make accounting (besides accounting on the PSTN
>> GW/Voice Switch) when those to identifiers are not globally uniq?
>>
>> So far ive seen non uniq callids from Grandstream Products
>> and non uniq
>> tags from sipura adapters.
>>
>> :: arnd ::
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
hello friends,
iam very much new to the sems
i have downloaded the sems from berlios.de
compiled and installed successfully
now my doubts are
1) is there any minimum requirements (hardware
,soundcard) are there for ser box to run the media
server sems
2) iam using the xlite as my user agents
so i think if one user is not logged in then the other
user agent can send his message in form of the .wav to
his mail
so how to use it practically in my case .
can i expect like this i.e if one user not logged in
then the other user still can record message and send
it the respective mail id
so explain me if possible please give me sample
ser.cfg file for sems
i have seen in the source tree the voicemail.cfg but
could not able understand
with regards
serdiehard
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Hi guys,
Im having a small problem with the PDT module. It seems to be installed
with the database set up correctly, and it seems to translate the codes
correctly. However, when a call is placed using the PDT module, the
invite appears to be incomplete (at least, as reported by acc module),
and reports:
ACC: transaction answered: method=INVITE,
i-uri=sip:839503018@sip.dev.inmarsat.com, o-uri=sip:3018@sip.prodec.tv
Eventually, SER gets a timeout.
I was wondering two things - does it look like this is indeed a
malformed invite or is that that my ser.cfg for some reason isn't
responding correctly to the reply from the forwarded domain?
Incidentally, if I dial the external address directly from the telephone
keypad, all is fine and I see the following in the syslog.
ACC: transaction answered: method=INVITE, i-uri=sip:3018@sip.prodec.tv,
o-uri=sip:3018@sip.prodec.tv,
call_id=62C50F87-E5D1-4F8B-B0BD-5A1146A4A812(a)161.30.94.150, from=Dave
Bath <sip:admin@sip.dev.inmarsat.com>;tag=2642545628, code=200
If anyone has any ideas, I have posted the ngreps underneath. If you
want to take a look at the ser.cfg, then please just let me know.
Thanks very much in advance,
Dave
---- Dialing Using pdt module .... Should the "to" field in the 3rd
trace be sip:839503018@sip.dev.inmarsat.com or should it be
3018(a)sip.prodec.tv now?
U 161.30.94.151:3837 -> 161.30.94.136:5060
INVITE sip:839503018@sip.dev.inmarsat.com SIP/2.0..From:
sip:test3@sip.dev.inmarsat.com;tag=1c24041..To: sip:8395
03018@sip.dev.inmarsat.com..Call-Id:
call-1092641177-25@161.30.94.151..Cseq: 1 INVITE..Contact:
<sip:test3@161.30
.94.151>..Content-Type: application/sdp..Content-Length:
308..Accept-Language: en..Allow: INVITE, ACK, CANCEL, BY
E, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc,
sip-cc-01, timer, replaces..User-Agent: Pingte
l/2.1.11 (VxWorks)..Date: Mon, 16 Aug 2004 07:26:23 GMT..Via:
SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5 IN
IP4 161.30.94.151..s=phone-call..c=IN IP4 161.30.94.151..t=0
0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap:96
eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0
pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=rtpmap:18 g729/8000/
1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1..
#
U 161.30.94.136:5060 -> 161.30.94.151:5060
SIP/2.0 100 trying -- your call is important to us..From:
sip:test3@sip.dev.inmarsat.com;tag=1c24041..To: sip:839
503018@sip.dev.inmarsat.com..Call-Id:
call-1092641177-25@161.30.94.151..Cseq: 1 INVITE..Via: SIP/2.0/UDP
161.30.9
4.151..Server: Sip EXpress router (0.8.14
(i386/linux))..Content-Length: 0..Warning: 392 161.30.94.136:5060 "Nois
y feedback tells: pid=15219 req_src_ip=161.30.94.151
req_src_port=3837 in_uri=sip:839503018@sip.dev.inmarsat.com
out_uri=sip:3018@sip.prodec.tv. via_cnt==1"....
#
U 161.30.94.136:5060 -> 80.234.135.99:5060
INVITE sip:3018@sip.prodec.tv. SIP/2.0..Max-Forwards:
10..Record-Route: <sip:839503018@161.30.94.136;ftag=1c24041
;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=1c24041..To:
sip:839503018@sip.dev.inmarsat.com..Call-Id: call-
1092641177-25@161.30.94.151..Cseq: 1 INVITE..Contact:
<sip:test3@161.30.94.151>..Content-Type: application/sdp..C
ontent-Length: 308..Accept-Language: en..Allow: INVITE, ACK, CANCEL,
BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSC
RIBE..Supported: sip-cc, sip-cc-01, timer, replaces..User-Agent:
Pingtel/2.1.11 (VxWorks)..Date: Mon, 16 Aug 2004
07:26:23 GMT..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bKc80d.2fb31ae5.0..Via: SIP/2.0/UDP
161.30.94.151....v=
0..o=Pingtel 5 5 IN IP4 161.30.94.151..s=phone-call..c=IN IP4
161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0
8 18 98..a=rtpmap:96 eg711u/8000/1..a=rtpmap:97
eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=
rtpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98
telephone-event/8000/1..
---- Dialling directly
U 161.30.94.150:5060 -> 161.30.94.136:5060
INVITE sip:3018@sip.prodec.tv SIP/2.0..Via: SIP/2.0/UDP
161.30.94.150:5060;rport;branch=z9hG4bKA9C83C1BE5274A2DB6
11818F87419F33..From: Dave Bath
<sip:admin@sip.dev.inmarsat.com>;tag=2642545628..To:
<sip:3018@sip.prodec.tv>..Co
ntact: <sip:admin@161.30.94.150:5060>..Call-ID:
62C50F87-E5D1-4F8B-B0BD-5A1146A4A812@161.30.94.150..CSeq: 34989 I
NVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent:
X-Lite release 1103m..Content-Length: 298....
v=0..o=admin 945333497 945333507 IN IP4 161.30.94.150..s=X-Lite..c=IN
IP4 161.30.94.150..t=0 0..m=audio 8000 RTP/
AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8
pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rt
pmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..
#
U 161.30.94.136:5060 -> 161.30.94.150:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
161.30.94.150:5060;rport=5060;branch=z9hG4bK
A9C83C1BE5274A2DB611818F87419F33..From: Dave Bath
<sip:admin@sip.dev.inmarsat.com>;tag=2642545628..To: <sip:3018@
sip.prodec.tv>..Call-ID:
62C50F87-E5D1-4F8B-B0BD-5A1146A4A812@161.30.94.150..CSeq: 34989
INVITE..Server: Sip EXpr
ess router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
161.30.94.136:5060 "Noisy feedback tells: pid=
15971 req_src_ip=161.30.94.150 req_src_port=5060
in_uri=sip:3018@sip.prodec.tv out_uri=sip:3018@sip.prodec.tv via
_cnt==1"....
#
U 161.30.94.136:5060 -> 80.234.135.99:5060
INVITE sip:3018@sip.prodec.tv SIP/2.0..Record-Route:
<sip:3018@161.30.94.136;ftag=2642545628;lr=on>..Via: SIP/2.0
/UDP 161.30.94.136;branch=z9hG4bK598a.f4d0fef2.0..Via: SIP/2.0/UDP
161.30.94.150:5060;rport=5060;branch=z9hG4bKA9
C83C1BE5274A2DB611818F87419F33..From: Dave Bath
<sip:admin@sip.dev.inmarsat.com>;tag=2642545628..To: <sip:3018@si
p.prodec.tv>..Contact: <sip:admin@161.30.94.150:5060>..Call-ID:
62C50F87-E5D1-4F8B-B0BD-5A1146A4A812(a)161.30.94.15
0..CSeq: 34989 INVITE..Max-Forwards: 69..Content-Type:
application/sdp..User-Agent: X-Lite release 1103m..Content
-Length: 298....v=0..o=admin 945333497 945333507 IN IP4
161.30.94.150..s=X-Lite..c=IN IP4 161.30.94.150..t=0 0..m
=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8
pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98
iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
Have you used the WinStun program to confirm that your stun server is
correctly set up?
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of ser die
Sent: 16 August 2004 08:44
To: serusers(a)lists.iptel.org
Subject: [Serusers] ser + stun
hello friends,
iam using ser of latest cvs head
and stun server of 0.92 version
iam running ser + stun on the same box
mines problem is stun could able to communicate
with the behind nat boxes and give an public ip to
that machines but ser could not able to even
detect that.
is there any special cofigurations we need to make
with regards
rama kanth
__________________________________________________
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_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
hello friends,
iam using ser of latest cvs head
and stun server of 0.92 version
iam running ser + stun on the same box
mines problem is stun could able to communicate
with the behind nat boxes and give an public ip to
that machines but ser could not able to even
detect that.
is there any special cofigurations we need to make
with regards
rama kanth
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
in the set-up
UAC(A)----------------SER-------------UAC(B)
if A calls B and then A hangs up all is fine,
if A calls B and then B hangs up a 481 Call leg/transaction error
occurs, this does not even get sent to the A client, the 481 is
generated by SER. B then thinks it is still in a call.
Should BYE messages be relayed? because it seems the server does not
relay any BYE messages at all (even the ones that work) according to
tcpdump.
Using SER 0.8.14 in record_route (this is a requirement) and MS RTC
based clients.
Thanks
--
Andrew Mee
