Hello!
We've a problem with a provisional response. The invite is sent through
NAT , and the response will be sent to incorrect port.
The invite is sent from 193.77.152.140:53559 to ser, (it is a natted
firewall. It not insert Via header, don't understand sip)
The topmost via is:
Via: SIP/2.0/UDP
10.10.1.2:5080;branch=z9hG4bK-a47f2525b20ab1a953a7c74174551819;received=193.77.152.140
The server recognize, that it is received from 193.77.152.140, but the
reply will be sent to 193.77.152.140:5080
according to the RFC 3261 the port used in the response is , what
present in the via header.
How can i force that the response will be sent to the source port
(52559)?
Thanks,
Tamas
Hi,
I have found the problem. (in fact it was not a problem)
The client was using the alternate media connection information which is peer-2-peer.
How can I prevent P2P connection ?
I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 6:13 PM
To: serusers(a)iptel.org
Subject: RE: [Serusers] Mediaproxy problem
Hi,
Please help me with the following problem.
Do I need SRV dns records even if I don't use load balancing ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] Mediaproxy problem
Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
Hi,
Please help me with the following problem.
Do I need SRV dns records even if I don't use load balancing ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] Mediaproxy problem
Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
Hi,
I have found the problem. (in fact it was not a problem)
The client was using the alternate media connection information which is peer-2-peer.
How can I prevent P2P connection ?
I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 6:13 PM
To: serusers(a)iptel.org
Subject: RE: [Serusers] Mediaproxy problem
Hi,
Please help me with the following problem.
Do I need SRV dns records even if I don't use load balancing ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] Mediaproxy problem
Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
Hi,
Please help me with the following problem.
Do I need SRV dns records even if I don't use load balancing ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] Mediaproxy problem
Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
<http://387555.sigclick.mailinfo.com/sigclick/03090E00/07074E03/00090C49/002…>
_____________________________________________________________________________________________________________________________________________
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Thanks Prashant,
Can you give me more details about the CDR tool that comes with
Asterisk. My final aim is to provide PC-to-Phone service on a small
scale. Will your solution help me.
Note that I am replying from a new email address
Regards
Concy
-----Original Message-----
From: Prashant Bhalesain [mailto:prashb2000@yahoo.com]
Sent: Monday, June 19, 2006 11:18 PM
To: Benjamin.George(a)t-systems.com; greger(a)teigre.com;
CPEREIRA(a)Human-soft.com
Cc: serusers(a)lists.iptel.org
Subject: RE: [Serusers] How to start a VOIP business ?
Boils down to Asterisk+SER I guess.
You can use asterisk as a voicemail server+media
server+CDR and SER as a SIP router. (connection to
PSTN/termination gateway?)
Asteriskathome provides most software bundled. worth
giving a try.
Prashant
--- Benjamin.George(a)t-systems.com wrote:
> Hi,
>
>
>
> I too can provide you some inputs on this apart from
> the things which are already mentioned:
>
>
>
> 1. You may require a media server for playing
> prompts, announcements, etc
> 2. Voice mail server to store and play back voice
> mails
> 3. Signaling + Media Gateway if you want to connect
> to PSTN world
> 4. Any good performance management/monitoring tool
> (normally part of OSS) to measure and manage the
> performance of the entire infrastructure to assure
> zero/minimum downtime
>
>
>
> Regards,
>
> Benjamin.
>
>
>
>
>
> _____
>
> From: Greger V. Teigre [mailto:greger@teigre.com]
> Sent: Monday, June 19, 2006 1:51 PM
> To: Concy Pereira
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] How to start a VOIP business
> ?
>
>
>
> It's a fairly big questions you're asking. But
> basically you're onto it ;-)
> Ad 1. You should go for something newer than 9.0
> Ad 2. You need to decide on your maintenance
> schedule and policies for upgrades/patching etc.
> Many larger-scale deployments are based on SER 0.9.x
> because a) it has proven extremely stable b) it is
> currently maintained for bug and security fixes and
> will be for a while. Others choose openser to get
> more features, but you need to make sure that you
> can handle testing and deployment of new versions
> more often.
> Ad 3. You need a RADIUS server with a user database
> or you can use mysql (you need mysql regardless of
> using RADIUS or not)
>
> g-)
>
>
> Concy Pereira wrote:
>
> Sir,
>
>
>
> I would like to know how can start a successful SIP
> base VOIP business. I have being going through some
> documentation of Open server software but could not
> come to a conclusion.
>
>
>
> Anybody can suggest me a complete solution for
> running PC to Phone service Using GPL products, with
> accounting, authentication and authorization with
> different rate Plans etc.
>
>
>
> Here is what I have decided to do.
>
>
>
> SIP Server (Hardware)
>
>
>
> 1. Linux RedHat 9.0
>
> 2. OpenSer or SIP express router
>
> 3. Radius client
>
>
>
> Accounting /CDR server ( Hardware)
>
>
>
> 1. Linux Redhat 9.0
>
> 2. CDRTool by ag-projects
> http://www.ag-projects.co
> <http://www.ag-projects.co> m/cdrtool.html
>
> 3. MySQL
>
>
>
> 1.Carrirer
>
>
>
> Subcription to any International whole sale VOIP
> carrier
>
>
>
> Please suggest if I have missed anything out.
>
>
>
> Regards,
>
>
>
> Concy
>
>
>
>
>
>
>
> _____
>
>
>
>
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> <mailto:Serusers@lists.iptel.org>
> http://lists.iptel.org/mailman/listinfo/serusers
> <http://lists.iptel.org/mailman/listinfo/serusers>
>
> > _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
On 06/19/06 20:42, Kelvin Williams wrote:
>
> My OpenSER box receives an Invite to
> sip:+11234567890@ip.add.re.ss:5060 – my XPro is registered as
> Kelvin(a)my.domain.com <mailto:Kelvin@my.domain.com>.
>
> When I receive this call, to my number 11234567890 – how do I make my
> XPro ring?
>
If you have only one account, then you can do it from the script:
rewriteuser("Kelvin");
lookup("location");
t_relay();
If you have many, you better use aliases. So just point 11234567890 to
kelvin in aliases table (see alias_db module
http://openser.org/docs/modules/1.0.x/alias_db.html) and then perform
user location lookup (see registrar/usrloc modules). It will translate in:
alias_db_lookup("dbaliases");
lookup("location");
t_relay();
Cheers,
Daniel
> kw
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
My OpenSER box receives an Invite to sip:+11234567890@ip.add.re.ss:5060 - my
XPro is registered as Kelvin(a)my.domain.com.
When I receive this call, to my number 11234567890 - how do I make my XPro
ring?
kw
Hi,
I want my SER(0.9.3) to store status and x, y coordinates into database and
to generate a NOTIFY whenever the position changes.
In the log-file, I can see this information, but SER doesn't store it in the
database and it doesn't generate a NOTIFY.
In the log file, I get the message: publish presentity: "No tuple for ....".
I hope, someone can help me.
Regards,
Alois
/usr/sbin/ser[25182]: handle publish request
/usr/sbin/ser[25182]: get_pres_uri: _puri=sip:PDA_User@fhjoanneum.at
/usr/sbin/ser[25182]: handle_publish -4- p_uri=
p_uri.len=23
/usr/sbin/ser[25182]: handle_publish -5- presentity=0x4069c0f8
/usr/sbin/ser[25182]: parse_pidf:
sipuri=0x821b230:sip:PDA_User@fh-joanneum.at contact=(nil):(null)
basic=0x821b2c8:open location=(nil):(null)
/usr/sbin/ser[25182]: parse_pidf: site=(nil):(null) floor=(nil):(null)
room=(nil):(null)
/usr/sbin/ser[25182]: parse_pidf: x=0x821be28:01524.3739
y=0x821be38:04704.0970 radius=(nil):(null)
/usr/sbin/ser[25182]: publish_presentity: no tuple for
PDA_User(a)fh-joanneum.at
/usr/sbin/ser[25182]: publish_presentity: event_package=1 -1-
/usr/sbin/ser[25182]: handle_publish -8- paerrno=0
/usr/sbin/ser[25182]: handled publish
PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP
10.0.0.3:5060;branch=z9hG4bK1443248938..Max-Forwards: 70..From:
<sip:PDA_User@fh-joanneum.at>;tag=271646917..To:
<sip:PDA_User@fh-joanneum.at>..Call-ID:
1040576725-1127390674-656438565..CSeq: 5 PUBLISH..Contact:
<sip:PDA_User@fh-joanneum.
at>..Content-Type: application/cpim-pidf+xml..Content-Length:
308..Expires:7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation
version, www.independentsoft.com....<?xml version="1.0" encoding="utf-8"?>
<presencexmlns="urn:ietf:params:xml:ns:pidf"entity="sip:PDA_User@fh-joanneum.at">
<tupleid="9r28r59">
<status>
<basic>open</basic>
<geopriv><locationinfo><civilAddress><x>01524.3595</x><y>04704.0990</y></civilAddress></locationinfo></geopriv>
</status>
</tuple>
</presence>..
SIP/2.0 200 OK..Via: SIP/2.0/UDP
10.0.0.3:5060;branch=z9hG4bK1443248938..From:
<sip:PDA_User@fh-joanneum.at>;tag=271646917..To:
<sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4cd6..Call-ID:
1040576725-1127390674-656438565..CSeq: 5 PUBLISH..Server: Sip EXpress router
(0.9.3 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy
feedback tells:pid=25182 req_src_ip=10.0.0.3 req_src_port=5060
in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at
via_cnt==1"
Hi all,
i need to implement openser for nat traversal. can anyone tell me if using nat helper module would be better or using media proxy. from so far i read, i understood that mediaproxy needs separate installation but does not need to make much change in the existing openser.cfg. where as if i use the nathelper, there is no need for separate media proxy installation but the openser cfg needs to be changed. Please suggest the best way to do it....? also please correct me if what i understood is not right......
Thanks so much,
Padmaja
The client can't register, still it doesn't allow registration.
Attached is the ngrep output you mentioned.
Thanks,
Hamid
Ngrep Output:
#
U 85.15.7.17:3166 -> 85.15.7.18:5060
REGISTER sip:shatel.ir SIP/2.0..From: sip:101@shatel.ir;tag=5fa04040..To:
sip:101@shatel.ir..Call-Id: 9401fb256987b29da88b308b1
1ce64ed..Cseq: 1 REGISTER..Contact:
<sip:101@85.15.7.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>..Expires:
300..Date: Mon, 19 J
un 2006 11:02:02 GMT..Max-Forwards: 20..User-Agent: sipX/2.5.2
(WinNT)..Accept-Language: en..Via: SIP/2.0/UDP 85.15.7.17;branch
=z9hG4bK-a4ba80ab4920a96042a782f74c40944d..Content-Length: 0....
#
U 85.15.7.17:3166 -> 85.15.7.18:5060
REGISTER sip:shatel.ir SIP/2.0..From: sip:101@shatel.ir;tag=5fa04040..To:
sip:101@shatel.ir..Call-Id: 9401fb256987b29da88b308b1
1ce64ed..Cseq: 1 REGISTER..Contact:
<sip:101@85.15.7.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>..Expires:
300..Date: Mon, 19 J
un 2006 11:02:02 GMT..Max-Forwards: 20..User-Agent: sipX/2.5.2
(WinNT)..Accept-Language: en..Via: SIP/2.0/UDP 85.15.7.17;branch
=z9hG4bK-a4ba80ab4920a96042a782f74c40944d..Content-Length: 0....
#
U 85.15.7.17:3166 -> 85.15.7.18:5060
REGISTER sip:shatel.ir SIP/2.0..From: sip:101@shatel.ir;tag=5fa04040..To:
sip:101@shatel.ir..Call-Id: 9401fb256987b29da88b308b1
1ce64ed..Cseq: 1 REGISTER..Contact:
<sip:101@85.15.7.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>..Expires:
300..Date: Mon, 19 J
un 2006 11:02:02 GMT..Max-Forwards: 20..User-Agent: sipX/2.5.2
(WinNT)..Accept-Language: en..Via: SIP/2.0/UDP 85.15.7.17;branch
=z9hG4bK-a4ba80ab4920a96042a782f74c40944d..Content-Length: 0....
#
U 85.15.7.18:5060 -> 85.15.7.17:5060
SIP/2.0 401 Unauthorized..From: sip:101@shatel.ir;tag=5fa04040..To:
sip:101@shatel.ir;tag=a2c1e7c3808498f2cb2d193cd520e0fc.60d6
..Call-Id: 9401fb256987b29da88b308b11ce64ed..Cseq: 1 REGISTER..Via:
SIP/2.0/UDP 85.15.7.17;branch=z9hG4bK-a4ba80ab4920a96042a78
2f74c40944d..WWW-Authenticate: Digest realm="shatel.ir",
nonce="449677461ef22770698d74f5dde4b0c2f00cf44f"..Server: OpenSer (1.0
.1 (i386/linux))..Content-Length: 0..Warning: 392 85.15.7.18:5060 "Noisy
feedback tells: pid=13352 req_src_ip=85.15.7.17 req_s
rc_port=3166 in_uri=sip:shatel.ir out_uri=sip:shatel.ir via_cnt==1"....
#
U 85.15.7.18:5060 -> 85.15.7.17:5060
SIP/2.0 401 Unauthorized..From: sip:101@shatel.ir;tag=5fa04040..To:
sip:101@shatel.ir;tag=a2c1e7c3808498f2cb2d193cd520e0fc.60d6
..Call-Id: 9401fb256987b29da88b308b11ce64ed..Cseq: 1 REGISTER..Via:
SIP/2.0/UDP 85.15.7.17;branch=z9hG4bK-a4ba80ab4920a96042a78
2f74c40944d..WWW-Authenticate: Digest realm="shatel.ir",
nonce="4496774acfd59b6e20c1968fc0bff804d526ce03"..Server: OpenSer (1.0
.1 (i386/linux))..Content-Length: 0..Warning: 392 85.15.7.18:5060 "Noisy
feedback tells: pid=13352 req_src_ip=85.15.7.17 req_s
rc_port=3166 in_uri=sip:shatel.ir out_uri=sip:shatel.ir via_cnt==1"....
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
Sent: Monday, June 19, 2006 4:32 PM
To: Hamid Ali Asgari
Cc: users(a)openser.org
Subject: Re: [Users] SIP Clients Can't REGISTER
Hello,
On 06/19/06 14:38, Hamid Ali Asgari wrote:
> Hi,
> Thanks for the reply.
> I commented out the entry you mentioned:
>
> #101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended",
> #Service-Type == "Group-Check"
>
> What happens now is that the radius will check for a group named SUSPENDED
> and will reject because it won't match the 101 user:
>
> >From RADIUS debug:
> users: Matched entry DEFAULT at line 152
> users: Matched entry DEFAULT at line 255
> -----------------------------------------------------------
> Users:152: DEFAULT Auth-Type = System
> Fall-Through = 1
> Users:255: DEFAULT Auth-Type := Reject, Service-Type == "Group-Check"
>
> Any thoughts?
>
this is ok, it is the response that the user is not in the suspended
group. The processing should go further in openser configuration script.
Do you encounter other issues with calls/registrations?
Cheers,
Daniel
> Hamid
>
>
>
>
> -----Original Message-----
> From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
> Sent: Monday, June 19, 2006 12:54 PM
> To: Hamid Ali Asgari
> Cc: 'Edson'; users(a)openser.org
> Subject: Re: [Users] SIP Clients Can't REGISTER
>
> If look carefully in the configuration file of the tutorial, it checks
> for users belonging to "suspended" group. As you have in 'users' file:
>
> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended",
> Service-Type == "Group-Check"
>
> Reply-Message = "Authorized"
>
> means that user 101 is in suspended group and it is not allowed to
> register or initiate calls. So remove this line for your 'users' file.
>
> Also, when you test, just watch the network traffic on you sip server,
> it will reveal the reply code so you can spot easily which statement was
> hit in the config file: ngrep -d any port 5060
>
> Cheers,
> Daniel
>
>
Hi,
Please help me with the following problem.
Do I need SRV dns records even if I don't use load balancing ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] Mediaproxy problem
Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
Hi,
Please help me with the following problem.
Do I need SRV dns records even if I don't use load balancing ?
Thanks,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net)
Sent: Monday, June 19, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] Mediaproxy problem
Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
<http://387555.sigclick.mailinfo.com/sigclick/010C0605/07024D06/0203064C/007…>
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Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez.
This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence.
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Hi,
I just installed mediaproxy 1.7.2
At this time, I am not interested in using more than one rtp server for loadbalancing.
That's why I didn't add any DNS SRV records to my DNS.
Would that be a problem ?
If it's not a problem, then what's the problem described below?
When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server)
But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer)
Thanks for your comments,
ilker
<http://387555.sigclick.mailinfo.com/sigclick/0A090604/07074E00/0906054D/001…>
_____________________________________________________________________________________________________________________________________________
Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez.
This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence.
_____________________________________________________________________________________________________________________________________________