Sorry, but for security reasons I don't read other people's dumps.
You just need to check whether the m= line refers to your rtpproxy's
public IP and not the private. And of course that rtpproxy and
nathelper communicate with eachother...
g-)
> harry gaillac wrote:
>> I've built rtpproxy with -i option :
>>
>> root 14362 1 0 16:39 ? 00:00:00
>> /usr/local/bin/rtpproxy -l 192.168.0.1 -i
>> 80.119.11.112
>>
>> However I can't send or receive invite from/to user
>> agent .
>> look at invitation file (ethereal)!
>> Should I configure phone per port ?
>> What's wrong ?
>>
>> Harry
>>
>>
>> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit :
>>
>>> The patch was current for rtpproxy from repository
>>> at that point. I sent the
>>> patch to Maxim and asked if it could be included in
>>> CVS, but I never heard
>>> from him.
>>> If you add the patch by hand, you should be able
>>> to make rtpproxy
>>> announce the public IP in the INVITEs and OKs
>>> (instead of rtpproxy's
>>> private/interface address).
>>> g-)
>>>
>>> harry gaillac wrote:
>>>> I tried to patch main.c without success according
>>> to
>>>> main.c.rej but i try to add -i by hand.
>>>>
>>>> I failed to patch main.c .
>>>>
>>>> If I add -i options to rtpproxy will it solve
>>> problem
>>>> of signaling behind nat (on same box) ?
>>>>
>>>>
>>>> Harry
>>>>
>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit
>>>>
>>>>
>>>>>
>>>>
>>>
>> http://lists.iptel.org/pipermail/serusers/2005-January/014688.html
>>>>> harry gaillac wrote:
>>>>>> Where can I get your patch ?
>>>>>>
>>>>>> Harry
>>>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a
>>> écrit
>>>>>>
>>>>>>
>>>>>>> Running rtpproxy on a NATed box is a problem
>>> (you
>>>>>>> need a patch I posted a
>>>>>>> while back to announce the correct public IP).
>>>>>>> Running all daemons on the
>>>>>>> same server with a public IP is NOT a problem.
>>>>>>> g-)
>>>>>>> harry gaillac wrote:
>>>>>>>> Hello,
>>>>>>>>
>>>>>>>> SER+nathelper+rtpproxy run on the same box that nat.
>>>>>>>> Is it a problem ?
>>>>>>>>
>>>>>>>> Harry
>>>>>>>>
>>>>>>>>
>>>>>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a
>>>>> écrit
>>>>>>>>
>>>>>>>>
>>>>>>>>> That's strange. We are using some Polycom 300
>>>>> and
>>>>>>>>> they work ok. No STUN
>>>>>>>>> support and sometimes a very erratic
>>>>> registration
>>>>>>>>> pattern, but except from
>>>>>>>>> that, they seem to perform ok. They
>>>>> interoperate
>>>>>>>>> with sipuras,
>>>>>>>>> grandstreams, sjphone, cisco gw, xlite, etc.
>>>>>>>>> g-)
>>>>>>>>>
>>>>>>>>> Juha Heinanen wrote:
>>>>>>>>>> harry,
>>>>>>>>>>
>>>>>>>>>> we tried polycom video phones a month or so
>>> ago and their sip
>>>>>>>>>> implementation was worst i had seen so far.
>>> the phones didn't
>>>>>>>>>> support digest authentication and we were also hit the
>>>>>>>>>> problem you mentioned with domain in request-uri. i suggest
>>>>>>>>>> to junk those phones before their sip implementation is
>>>>>>>>>> closer to
>>> rfc3261.
>>>>>>>>>>
>>>>>>>>>> -- juha
>>>>>>>>>>
>>>>>>>>>>
>>>>> _______________________________________________
>>>>>>>>>> Serusers mailing list
>>>>>>>>>> serusers(a)lists.iptel.org
>>>>>>>>>>
>>>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>> _____________________________________________________________________________
>>>>>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage
>>>>>>>> pour vos mails, photos et vidéos ! Créez votre Yahoo!
>>> Mail sur
>>>>>>>> http://fr.mail.yahoo.com
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>> _____________________________________________________________________________
>>>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace
>>>>> de stockage pour vos
>>>>>> mails, photos et vidéos ! Créez votre Yahoo! Mail sur
>>>>>> http://fr.mail.yahoo.com
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>> _____________________________________________________________________________
>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace
>>> de stockage pour vos
>>>> mails, photos et vidéos ! Créez votre Yahoo! Mail
>>> sur
>>>> http://fr.mail.yahoo.com
>>>
>>>
>>
>>
>>
>>
>>
>>
>> _____________________________________________________________________________
>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos
>> mails, photos et vidéos ! Créez votre Yahoo! Mail sur
>> http://fr.mail.yahoo.com
Hello,
I've been trying to add support for ATAs behind NAT firewalls for
several days, and have had nothing but frustrations with it, so
hopefully someone can clue me in. I tried modifying my existing ser
config which has been working great, but failing that, I finally got the
stock config file from
ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.3.05.cfg,
modified my IP address for NAT and used that (after adding the domain
module).
My first problem is that when an ATA registers behinf the NAT firewall,
the NAT flag is never set, so if doesn't get the pings every 30 seconds.
I can manually set that flag by just having setflag(6) in the config
file and that seems to temporarily fix that problem, but I can't tell
that it does anything else.
Outbound calls from the ATA work just fine, which they always have even
where I had no NAT stuff in SER at all. Inbound calls to the ATA get
ignored my it though, this is a Cisco ATA-186. I hooked up an old hub
so I could get the messages using ngrep on the side with the NAT router
and the main server, and here's what I see. 7771111001 is the ATA,
7778881000 is a local phone. 11.11.11.100 is the SER router,
11.11.11.44 is the PSTN gateway initiating the call, 11.11.11.18 is
the NAT router's public IP, and 10.0.2.3 is the ATA.
SER router's ngrep:
U 11.11.11.100:5060 -> 11.11.11.18:1387 27067@0:1480
...k....INVITE
sip:7771111001@11.11.11.18:1387;user=phone;transport=udp SI
P/2.0..Record-Route: <sip:11.11.11.100;ftag=6B826304-1A6A;lr=on>..Via:
SIP/2
.0/UDP 11.11.11.100;branch=z9hG4bKeda1.2545.3..Via: SIP/2.0/UDP
11.11.11.44
:5060;x-route-tag="tgrp:lnx";branch=z9hG4bK8C9D..From:
<sip:7778881000@11.11
11.44>;tag=6B826304-1A6A..To: <sip:7771111001@11.11.11.100>..Date:
Thu, 26
May 2005 14:10:42 GMT..Call-ID:
C7AED903-CD2611D9-8130A2A3-EEBB2457(a)11.11.
11.44..Supported: 100rel,timer..Min-SE: 1800..Cisco-Guid:
3350084827-34418
24217-2183397385-1128595304..User-Agent:
Cisco-SIPGateway/IOS-12.x..Allow:
INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY,
INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards:
16..Remote-Party-ID
:
<sip:7778881000@11.11.11.44>;party=calling;screen=no;privacy=off..Timesta
mp: 1117116642..Contact: <sip:7778881000@11.11.11.44:5060>..Expires:
180..A
llow-Events: telephone-event..Content-Type:
application/sdp..Content-Length
: 506....v=0..o=CiscoSystemsSIP-GW-UserAgent 2462 1180 IN IP4
11.11.11.44..
s=SIP Call..c=IN IP4 11.11.11.44..t=0 0..m=audio 16536 RTP/AVP 18 2 0
100 1
01..c=IN IP4 11.11.11.44..a=rtpmap:18 G729/8000..a=fmtp:18
annexb=yes..a=rt
pmap:2 G726-32/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:100
X-NSE/8000..a=fmtp:
100 192-194,200-202..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=
X-sqn:0..a=X-cap: 1 audio RTP/AVP 100..a=X-cpar: a=rtpmap:100
X-NSE/8000..a
=X-cpar: a=fmtp:100 192-194,200-202..a=X-cap: 2 image u
NAT (ATA location) ngrep:
U 11.11.11.100:5060 -> 10.0.2.3:5060 27070@0:1462
.......&INVITE
sip:7771111001@11.11.11.18:1405;user=phone;transport=udp S
IP/2.0..Record-Route:
<sip:11.11.11.100;ftag=6B826304-1A6A;lr=on>..Via:
SIP
/2.0/UDP 11.11.11.100;branch=z9hG4bKeda1.2545.6..Via: SIP/2.0/UDP
11.11.11.44:5060;x-route-tag="tgrp:lnx";branch=z9hG4bK8C9D..From:
<sip:7778881000@
11.11.11.44>;tag=6B826304-1A6A..To:
<sip:7771111001@11.11.11.100>..Date: Th
u, 26 May 2005 14:10:42 GMT..Call-ID:
C7AED903-CD2611D9-8130A2A3-EEBB2457@
11.11.11.44..Supported: 100rel,timer..Min-SE: 1800..Cisco-Guid:
335008482
7-3441824217-2183397385-1128595304..User-Agent:
Cisco-SIPGateway/IOS-12.x.
.Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards:
16..Remot
e-Party-ID:
<sip:7778881000@11.11.11.44>;party=calling;screen=no;privacy=o
ff..Timestamp: 1117116642..Contact:
<sip:7778881000@11.11.11.44:5060>..Exp
ires: 180..Allow-Events: telephone-event..Content-Type:
application/sdp..C
ontent-Length: 478....v=0..o=CiscoSystemsSIP-GW-UserAgent 2462 1180 IN
IP4
11.11.11.44..s=SIP Call..c=IN IP4 11.11.11.44..t=0 0..m=audio 16536
RTP/A
VP 18 2 0 100 101..c=IN IP4 11.11.11.44..a=rtpmap:18
G729/8000..a=fmtp:18
annexb=yes..a=rtpmap:2 G726-32/8000..a=rtpmap:0
PCMU/8000..a=rtpmap:100 X-
NSE/8000..a=fmtp:100 192-194,200-202..a=rtpmap:101
telephone-event/8000..a
=fmtp:101 0-16..a=X-sqn:0..a=X-cap: 1 audio RTP/AVP 100..a=X-cpar:
a=rtpma
p:100 X-NSE/8000..a=X-cpar: a=fmtp:100 192-194,200-202..
It looks like the ATA is getting the message, but just flat out ignoring
it. Anyone else have this problem?
Brian
Yes.
I noticed that the variable used by the avp_load_radius("caller") to store
the answer from the RADIUS server is in fact the one i defined first (var1),
but is has a prefix called "caller_" , so the variable is finally :
"caller_var1". Jackpot!.
So i used this in my ser.cfg:
if( !avp_load_radius("caller")) {
xlog("L_INFO", "time [%Tf] FAIL AVP_LOAD_RADIUS!\n");
};
if( !avp_pushto("$ruri", "s:caller_var1/g")) {
xlog("L_INFO", "time [%Tf] FAIL avp_pushto()!\n");
};
Thanks Greg for your help.
Just one more thing. This could be indicated in the module's README, i
think is an important thing that a user should know.
Regards!!.
Ricardo Martinez.-
> -----Mensaje original-----
> De: Greger V. Teigre [mailto:greger@teigre.com]
> Enviado el: Viernes, 27 de Mayo de 2005 3:57
> Para: Ricardo Martinez; serusers(a)lists.iptel.org
> Asunto: Re: [Serusers] avp_radius and avpops
>
>
> Do the debugging dance, so that you can verify what the calls
> to avp_* do...
> g-)
>
> Ricardo Martinez wrote:
> > Hello Greg
> > Thanks for your answer.
> > I made the change but it seems not to do anything....
> > Can someone help me here?
> >
> > Thanks in advance
> >
> > Regards
> > Ricardo.-
> >
> >> -----Mensaje original-----
> >> De: Greger V. Teigre [mailto:greger@teigre.com]
> >> Enviado el: Miércoles, 25 de Mayo de 2005 16:24
> >> Para: Ricardo Martinez; serusers(a)lists.iptel.org
> >> Asunto: Re: [Serusers] avp_radius and avpops
> >>
> >>
> >>> if( !avp_pushto("$RURI", "s:var1/g")) {
> >> replace with:
> >> "s:$var1/g"
> >> I don't know about the case sensivtivity of destination, I
> >> always use lower
> >> case...
> >> g-)
> >>
> >> Ricardo Martinez wrote:
> >>> Hello List.
> >>> I have a question regarding the use of avp_radius and avpops. I'm
> >>> using avp_radius to obtain an AVP value from my database
> via radius.
> >>> What i what to do is replace this value for the RURI.
> >>> Here i have a couple of questions.
> >>> 1.- The value returned by the avp_radius (the SIP-AVP) where is
> >>> stored ? It suppose that the SIP-AVP returned by radius
> >> has the form
> >>> of "name:value". That "name" refers to the name of what?.
> >>> For example i'm returning :
> >> "var1:sip:1234567@mydomain.com". What i
> >>> see in the debug is :
> >>>
> >>> avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has
> been added
> >>>
> >>> This is what i got in my ser.cfg (a snippet).
> >>>
> >>> if (method=="INVITE" || method=="CANCEL") {
> >>> if( !avp_load_radius("caller")) {
> >>> log (1, "AVP_RADIUS: Fail on
> avp_radius\n");
> >>> };
> >>
> >>>
> >>> if( !avp_pushto("$RURI", "s:var1/g")) {
> >>> log (1, "AVPOPS: Fail on AVPOPS\n");
> >>> };
> >>> };
> >>>
> >>> Again the debug = 9 .
> >>>
> >>> 6(23815) avp_load_user: AVP
> >> 'var1'='sip:1234567@mydomain.com' has been
> >>> added
> >>> 6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c:
> >>> load_avp_user(344)
> >>> 6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from
> >>> avp_radius.c: load_avp_user(330)
> >>> 6(23815) DEBUG:avpops:pushto_avp: no avp found
> >>> 6(23815) AVPOPS: Fail on AVPOPS
> >>>
> >>> What i'm doing wrong?
> >>> Thanks!
> >>>
> >>> Regards
> >>> Ricardo Martinez
> >>>
> >>> _______________________________________________
> >>> Serusers mailing list
> >>> serusers(a)lists.iptel.org
> >>> http://lists.iptel.org/mailman/listinfo/serusers
>
Hi everybody,
I setup ser 0.8.14 with mysql support on fedora core 2. And two UA
could connect over SER successfully.
I give a static IP to my SIP PROXY and followed ONSIP instructions
about mediaproxy. But I could not register UAC's behind NAT.
begining of my route block:
if (method=="INVITE" && client_nat_test("3")) {
# INSERT YOUR IP ADDRESS HERE
record_route_preset("PROXY_IP:5060;nat=yes");
} else if (method!="REGISTER") {
record_route();
};
# -----------------------------------------------------------------
# Call Tear Down Section
# -----------------------------------------------------------------
if (method=="BYE" || method=="CANCEL") {
end_media_session();
};
# -----------------------------------------------------------------
# Loose Route Section
# -----------------------------------------------------------------
if (loose_route()) {
if (has_totag() && (method=="INVITE" || method=="ACK")) {
if (client_nat_test("3") ||
search("^Route:.*;nat=yes")) {
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
my register block:
# -----------------------------------------------------------------
# REGISTER Message Handler
# ----------------------------------------------------------------
sl_send_reply("100", "Trying");
if (!search("^Contact:\ +\*") && client_nat_test("7")) {
setflag(6);
fix_nated_contact();
force_rport();
};
if (!www_authorize("PROXY_IP","subscriber")) {
www_challenge("PROXY_IP","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
invite block:
# -----------------------------------------------------------------
# CANCEL and INVITE Message Handler
# -----------------------------------------------------------------
if (client_nat_test("3")) {
setflag(7);
force_rport();
fix_nated_contact();
};
lookup("aliases");
if (uri!=myself) {
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
if (method=="CANCEL") {
route(1);
break;
};
if (!proxy_authorize("PROXY_IP","subscriber")) {
proxy_challenge("PROXY_IP","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
if (isflagset(6) || isflagset(7)) {
use_media_proxy();
};
route(1);
Any help would be appreciated...
Is it possible to configure SER to use RTP proxy only for specific
clients, ie ones which have a particular username pattern?
I would like to proxy the media only for those phones I have which do
not support ICE but support STUN.
Hello All,
The following two sites appear to be SIP service providers:
www.freeworlddialup.com/voip.brujula.net/english/messenger-howto.html
But when I follow their instruction for setting up
Windiows Messenger 4.6, my login attempts tell me,
"The service provider is temporarly unavailable, try again later"
This makes me suspect that they have shut down their voip servers,
and simply failed to delete their web pages.
Does anybody know the status of these two companies?
They have not responded to my email queries.
Freeworlddialup provides a simple tool to evaluate your
internet connection before trying their service.
The tool consistently claims that my port 5060 is blocked,
even though I am directly connected to my cable modem
and have all firewalls turned off. It seems like this
false result could be caused by an attempt establish
a 5060 connection to a non-existant voip server,
again supporting my theory that their server has been
shut down. Does this theory make sense?
Also, both providers say that they work with
Windows Messenger 4.6 and 4.7 but not with 5.1.
Did Microsoft commit some crime in version 5.1
to make it incompatible
with third party SIP service providers?
thanks for your help,
Michael
Hi all,
I'm using SER 0.8.14, FreeRadius 1.0.2. and XLite 1103. My question is about
MD5, why I can't see Digest-Algorithm = "MD5" in FreeRadius and XLite client
outputs.
Is this a problem in XLite or SER? Everything is configured as in HOW TO and
I get authenticated with FreeRadius so I suppose that configuration is OK.
Dictionaries are the same in radiusclient on SER machine and in FreeRadius.
I can provide more outputs if someone is interested to help.
Any help would be appreciated.
*****This is output from FreeRadius Server****** (no Digest-Algorithm =
"MD5" received)
...
rad_recv: Access-Request packet from host 161.53.0.131:33854, id=131,
length=207
User-Name = "djovanovic.srce"
Digest-Attributes = 0x0a11646a6f76616e6f7669632e73726365
Digest-Attributes = 0x0109737263652e6872
Digest-Attributes =
0x022a3432393439336464646235643930376433653433323765333063366566646566336436
6137643933
Digest-Attributes = 0x040d7369703a737263652e6872
Digest-Attributes = 0x030a5245474953544552
Digest-Response = "263ec6802b382ddf8d58b363fd0629ad"
Service-Type = Sip-Session
Sip-Uri-User = "djovanovic.srce"
NAS-IP-Address = 161.53.0.131
NAS-Port = 5060
Processing the authorize section of radiusd.conf
modcall: entering group authorize for request 0
modcall[authorize]: module "preprocess" returns ok for request 0
modcall[authorize]: module "chap" returns noop for request 0
modcall[authorize]: module "mschap" returns noop for request 0
rlm_digest: Converting Digest-Attributes to something sane...
Digest-User-Name = "djovanovic.srce"
Digest-Realm = "srce.hr"
Digest-Nonce = "429493dddb5d907d3e4327e30c6efdef3d6a7d93"
Digest-URI = "sip:srce.hr"
Digest-Method = "REGISTER"
rlm_digest: Adding Auth-Type = DIGEST
modcall[authorize]: module "digest" returns ok for request 0
...
*****This is output from XLite****** (no Digest-Algorithm = "MD5" sent)
SEND TIME: 203074718
SEND >> 191.153.206.58:5060
INVITE sip:537@srce.hr SIP/2.0
Via: SIP/2.0/UDP
191.153.206.57:5060;rport;branch=z9hG4bK8BB5DE10CC404EEBABE6761213B4AA45
From: davor <sip:djovanovic.srce@srce.hr>;tag=2361756632
To: <sip:537@srce.hr>
Contact: <sip:djovanovic.srce@191.153.206.57:5060>
Call-ID: 146A8327-BCC6-42E6-A027-83EA1832631B(a)191.153.206.57
CSeq: 40898 INVITE
Authorization: Digest
username="djovanovic.srce",realm="srce.hr",nonce="429492e156f0af6f7ae5e821d7
3df9be3e04f360",response="17a54d30cc2d5ecf1cc8e714d40210e0",uri="sip:537@src
e.hr"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 308
I am using SER for VOIP and experiencing issues with call setup when
making VOIP to VOIP calls. With 2 end units connected to the same
server, there is a 5-7 second delay in the RTP stream when I place a
call from one IP phone to another. I also experience a delay in ring
back when calling inbound to an IP phone from a standard PSTN line. I
am new to configuring SER. I can provide more info if needed. Thanks.
James D. Peters
Network Engineer
NetCarrier, Inc.
www.netcarrier.com <http://www.netcarrier.com>
