sr-users November 2005

sr-users@lists.kamailio.org

245 participants
570 discussions

[Serusers] Urgent NAT question: mediaproxy,rtp or STUN ?

by Adrian Georgescu

Antonello, There is no better solution for all possible cases. The solutions have differences explained in the readme files for example RTP proxy can take more load per server, Media Proxy can distribute load on multiple boxes based on DNS SRV records and STUN can help your Stun enabled clients to traverse the NAT in majority of cases but not all. So a combination between STUN plus one of the relay solutions will cover all cases. Configuring per session when to use Stun or a media relay is quite a daunting task given the number of possible call flow scenarios so if you wish that your calls always work regardless of UA type or call flow you will enable a media relay because Stun is not reliable 100%. Adrian >>>>>>>>>> Terrible question: Is mediaproxy better than rtp or STUN server for NAT problem ??? Radius Authentication work fine with Mediaproxy, STUN and RTP ? In Mediaproxy Server, radius auth supported only in non-free version (7.000,00 euro) ! My work is: Private Network for more than 1000 users Public SER SERVER Public Radius+MySQL Public Patton MediaGW Please, answer ! Thanks Antonello

19 years

[Serusers] Multiple entries in Location Table with Same Call-Id.

by Ramu Yadav

Hi All, With Some User-Agents we are facing multiple entries in "location" Table. The main problem is each and every time UA sending register packet with different port number in the contact uri. I am here with attaching the location table values as well as traces. Thanks -- Ramu Yadav

19 years

[Users] How Effective is STUN?

by Tavis P

I'm trying to find some statistics as to what the ratio of Cone vs Symmetric NAT solutions deployed in the world are, has anyone done some research into this? I'm curious what percentage of users in certain demographics (broadband clients, for example) i can expect to be serviced using STUN alone, so i can come up with some figure to help me build out my network Even just some anecdotal information of peoples experiences would be very useful Tavis

19 years

Re: [Users] Problem with the registration of X-PRO client with Openser

by Lokesh Kumar

----- Original Message ----- From: "Lokesh Kumar" <lokesh(a)interacesso.pt> To: <daniel(a)voice-system.ro> Sent: Friday, November 18, 2005 5:35 PM Subject: Re: [Users] Problem with the registration of X-PRO client with Openser > Hello Daniel, > > Thanks for your reply. > > Here is the ouput I got wheb i did ngrep on openser > > "----------snip---------------## > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > 192.168.0.47:5060;rport;branch=z9hG4bKA4 > > E96E15E966428C88FB407711BD924E..From: Alexandre Alves > <sip:3501@192.168.0.60>..To: Alexandre > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > <sip:3501@192.168.0.47:5060>..Cal > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 26879 > REGISTER..Expires: 1800..Ma > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > # > > I 192.168.0.60 -> 192.168.0.47 3:3 > > ....E.........1D.../...<......"MREGISTER sip:192.168.0.60 SIP/2.0..Via: > SIP/2.0/UDP 192.168. > > 0.47:5060;rport;branch=z9hG4bKA4E96E15E966428C88FB407711BD924E..From: > Alexandre Alves <sip:3 > > 501(a)192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: > "Alexandre Alves" < > > sip:3501@192.168.0.47:5060>..Call-ID: > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 2 > > 6879 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO build > 1101..Content-Length > > : 0....------------------snip-------------------------" > > > > BELOW IS THE OUTPUT I GOT WHEN I DO NGREP 5060...I AM UNABLE TO FIGURE OUT > ,WHATS WRONG WITH IT. PLEASE HELP. > > "------------------snip-------------- > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > 192.168.0.47:5060;rport;branch=z9hG4bK7E > > 650EAFB35F40FEB93391F06D534AC5..From: Alexandre Alves > <sip:3501@192.168.0.60>..To: Alexandre > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > <sip:3501@192.168.0.47:5060>..Cal > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 26883 > REGISTER..Expires: 1800..Ma > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > # > > I 192.168.0.60 -> 192.168.0.47 3:3 > > ....E.........0..../...<.......>REGISTER sip:192.168.0.60 SIP/2.0..Via: > SIP/2.0/UDP 192.168. > > 0.47:5060;rport;branch=z9hG4bK7E650EAFB35F40FEB93391F06D534AC5..From: > Alexandre Alves <sip:3 > > 501(a)192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: > "Alexandre Alves" < > > sip:3501@192.168.0.47:5060>..Call-ID: > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 2 > > 6883 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO build > 1101..Content-Length > > : 0....------------------snip-----------------------" > > ----- Original Message ----- > From: "Daniel-Constantin Mierla" <daniel(a)voice-system.ro> > To: "Lokesh Kumar" <lokesh(a)interacesso.pt> > Cc: <users(a)openser.org> > Sent: Friday, November 18, 2005 4:49 PM > Subject: Re: [Users] Problem with the registration of X-PRO client with > Openser > > > > It seems that the REGISTER message was not received by openser at all, > > otherwise a reply should be received. Do the ngrep on openser machine to > > see if the sip messages reach the system. Also, check if you have any > > firewall on port 5060 in place on openser server. > > > > Cheers, > > Daniel > > > > On 11/18/05 18:20, Lokesh Kumar wrote: > > > Hello Friends, > > > > > > I am having trouble with registration of X-PRO softphone with openser, > > > where my softphone is on 192.168.0.47 and openser is on 192.168.0.60. > > > My softphone says "Login Timed out , Contact Network admin". > > > > > > Here I am posting ngrep output also.Openser is having default configs, > > > but I had added this softphone as a user also. > > > > > > Can anyone help me please. > > > > > > Best Regards > > > Lokesh > > > > > > > > > [root@virtualmachine configs]# ngrep 192.168.0.47 > > > interface: eth0 (192.168.0.0/255.255.255.0) > > > match: 192.168.0.47 > > > ##################################################################### > > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > > > 192.168.0.47:5060;rport;branch=z9hG4bK96 > > > DBF3C437034F28A50968410DEB564C..From: Alexandre Alves > > > <sip:3501@192.168.0.60>..To: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > > > <sip:3501@192.168.0.47:5060>..Cal > > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 26821 > > > REGISTER..Expires: 1800..Ma > > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > > # > > > I 192.168.0.60 -> 192.168.0.47 3:3 > > > ....E...o\....H..../...<......<PREGISTER sip:192.168.0.60 > > > SIP/2.0..Via: SIP/2.0/UDP 192.168. > > > > > > 0.47:5060;rport;branch=z9hG4bK96DBF3C437034F28A50968410DEB564C..From: > > > Alexandre Alves <sip:3 > > > 501(a)192.168.0.60>..To <mailto:501@192.168.0.60%3E..To>: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" < > > > sip:3501@192.168.0.47:5060>..Call-ID: > > > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 2 > > > 6821 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO > > > build 1101..Content-Length > > > : 0.... > > > ##### > > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > > > 192.168.0.47:5060;rport;branch=z9hG4bK96 > > > DBF3C437034F28A50968410DEB564C..From: Alexandre Alves > > > <sip:3501@192.168.0.60>..To: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > > > <sip:3501@192.168.0.47:5060>..Cal > > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 26821 > > > REGISTER..Expires: 1800..Ma > > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > > # > > > I 192.168.0.60 -> 192.168.0.47 3:3 > > > ....E...o]....H..../...<......<PREGISTER sip:192.168.0.60 > > > SIP/2.0..Via: SIP/2.0/UDP 192.168. > > > > > > 0.47:5060;rport;branch=z9hG4bK96DBF3C437034F28A50968410DEB564C..From: > > > Alexandre Alves <sip:3 > > > 501(a)192.168.0.60>..To <mailto:501@192.168.0.60%3E..To>: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" < > > > sip:3501@192.168.0.47:5060>..Call-ID: > > > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 2 > > > 6821 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO > > > build 1101..Content-Length > > > : 0.... > > > ##exit > > > 78 received, 0 dropped > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

19 years

Fwd: [Serusers] mediaproxy module: RTP packet mapping after reINVITE

by Adrian Georgescu

Hi Joel, MediaProxy development goes further. I will take a look at your observations and let you know if there is any duplicate effort. Thanks Adrian >>>>>>>>>> Anyone have a comment on the problem that I observe in which the mediaproxy receives a request for a session with a media description that has a new (rather than additional) connection address, but does not act upon it? Is there continuing work going on for this module? Here is my latest code which seems to solve the problem of a simple 1-audio-session call, while attempting to preserve the existing video add-on capability. Please excuse the clumsiness of the code as I am new to python. While it doesn't solve the general SDP management case, it does solve the simple and most common case of 1 audio stream. If the project leadership thinks this is useful, I can come up with a more complete solution and submit the code. However, if there is already ongoing work in this area, I don't want to duplicate the effort.

19 years

[Serusers] Mediaproxy on Multiprocessor System

by Adrian Georgescu

Your RTP generator process is also using CPU, this is why you see both loaded during the test. MediaProxy uses only one CPU. Adrian >>>>>> Hi All, I run mediaproxy on a 2 CPUs machine (Fedora Code 3). When I stress the mediaproxy with rtpgenerator locally, i.e., just rtpgenerator.py --count=50, I noticed that the load is distributed evenly between the 2 CPUs. But when I stress remotely, using a seperate PC, i.e., rtpgenerator.py --ip=x.x.x.x --proxy=y.y.y.y --count=50, where x.x.x.x is rptgenerator's ip and y.y.y.y is mediaproxy's ip, I noticed that the load is mainly taken by CPU0.

19 years

Fwd: [Serusers] Get alias for user

by Voipers Portugal

Why don't you use rewritehost to do that? Jose Simoes 2005/11/19, Pol <kuroki(a)gmail.com>: > I Jan, I tryied your solution( creating a reversealiases table and do > lookup() function before rewrite host, but it didn't success. > > I have no idea on how to acomplish that!!! it seems to be easy but, remains > kicking me hard, and I'm stopped at this point, for several weeks. > I just want to forward incoming calls that are for users not On-line to > asterisk by replacing TO header for his/her alias. > > > INVITE sip:pnovell@domain.com ---> SER ( USER NOT ONLINE ) > | > | > lookup("aliases") pnovell=17940 > | > > ----> FORWARD sip:17940@IP_OF_ASTERISK > Any help? > > -- > Pol Novell > UPF- Universitat Pompeu Fabra. > > > On 10/24/05, Jan Janak <jan(a)iptel.org> wrote: > > > > On 24-10-2005 14:12, Damien Sandras wrote: > > > Le lundi 24 octobre 2005 ? 14:06 +0200, Samuel Osorio Calvo a écrit : > > > > lookup("aliases"); > > > > > > > > will do the job for you if aliases is the name of the table containing > the alias bindings (by default it will be). > > > > > > > > > > I had understood this, but the problem here is that I need the reverse: > > > the original request uri contains the username and I need to rewrite it > > > into the user alias. > > > > > > Example, SER receives INVITE dsandras @ gnomemeeting.net, and I want > > > to use rewriteuri to rewrite the INVITE into INVITE 600001 @ > > > gnomemeeting.net. > > > > > > Is that possible? > > > > Just setup reverse aliases. You can either use aliases table or create > > your own, let's say revaliases. > > > > Revaliases should have exactly same structure as aliases table and you > > should als add a row in version table for it. revaliases could contain > > usernames and map them to numbers. That is username would be dsandras > > and contact sip:600001@gnomemeeting.net . You can then call > > lookup("realiases") in the script. > > > > Jan. > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > _____________________________________________________________________ > > Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

19 years

[Users] Re: Users Digest, Vol 6, Issue 50

by Michael Kahlke

Tavis, Check out http://www.brynosaurus.com/pub/net/p2pnat/, it's not quite what you're asking for but it does include a table of data gathered from volunteers, detailing the percentage of NATs that support udp/tcp hole-punching and hairpin (loopback) translation. I'm concerned about the latter, it appears that not even ICE can resolve that issue. Mike > Date: Fri, 18 Nov 2005 15:36:35 -0800 > From: Tavis P <tavis.lists(a)galaxytelecom.net> > Subject: [Users] How Effective is STUN? > To: users(a)openser.org, serusers(a)iptel.org > Message-ID: <437E6583.5050905(a)galaxytelecom.net> > Content-Type: text/plain; charset=ISO-8859-1 > > I'm trying to find some statistics as to what the ratio of Cone vs > Symmetric NAT solutions deployed in the world are, has anyone done some > research into this? > > I'm curious what percentage of users in certain demographics (broadband > clients, for example) i can expect to be serviced using STUN alone, so i > can come up with some figure to help me build out my network > > Even just some anecdotal information of peoples experiences would be > very useful > > Tavis >

19 years

Re: [Serusers] Get alias for user

by Damien Sandras

Le lundi 24 octobre 2005 à 14:06 +0200, Samuel Osorio Calvo a écrit : > lookup("aliases"); > > will do the job for you if aliases is the name of the table containing the alias bindings (by default it will be). > I had understood this, but the problem here is that I need the reverse: the original request uri contains the username and I need to rewrite it into the user alias. Example, SER receives INVITE dsandras @ gnomemeeting.net, and I want to use rewriteuri to rewrite the INVITE into INVITE 600001 @ gnomemeeting.net. Is that possible? -- _ Damien Sandras (o- GnomeMeeting: http://www.gnomemeeting.org/ //\ FOSDEM 2005 : http://www.fosdem.org v_/_ H.323 phone : callto:ils.seconix.com/dsandras@seconix.com

19 years

[Serusers] Contact field in SIP HF between asterisk + ser

by harry gaillac

Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years

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sr-users November 2005