Hello,
Is it possible to do NAT traversal without RTP-PROXY ?
If yes, have you an example?
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Or is he simply asking about nathelper?
Mark
At 10:21 p.m. 20/02/2009, you wrote:
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
I will try to install my stun server.
If I install my stun server, when I will do save("location"), it will save the good IP and port in the db?
Cordialement, BERGANZ François
http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 20 février 2009 10:22 À : BERGANZ François Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
On Fri, Feb 20, 2009 at 9:40 AM, BERGANZ François < francois@acropolistelecom.net> wrote:
I will try to install my stun server.
If I install my stun server, when I will do save("location"), it will save the good IP and port in the db?
yes, because UAs will know their public IPs using STUN.
Thanks,
Asim
Cordialement, BERGANZ François
http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 20 février 2009 10:22 À : BERGANZ François Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
On Fri, Feb 20, 2009 at 9:48 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:40 AM, BERGANZ François < francois@acropolistelecom.net> wrote:
I will try to install my stun server.
If I install my stun server, when I will do save("location"), it will save the good IP and port in the db?
yes, because UAs will know their public IPs using STUN.
you can also use NATHELPER Module in KAMAILIO to fix NAT e.g
if(nat_uac_test("1")) { fix_nated_register();
Thanks,
Asim
Cordialement, BERGANZ François
http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 20 février 2009 10:22 À : BERGANZ François Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
On Fri, Feb 20, 2009 at 9:53 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:48 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:40 AM, BERGANZ François < francois@acropolistelecom.net> wrote:
I will try to install my stun server.
If I install my stun server, when I will do save("location"), it will save the good IP and port in the db?
yes, because UAs will know their public IPs using STUN.
you can also use NATHELPER Module in KAMAILIO to fix NAT e.g
if(nat_uac_test("1")) { fix_nated_register();
fix_nated_contact();
force_rport();
}
Thanks,
Asim
Cordialement, BERGANZ François
http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 20 février 2009 10:22 À : BERGANZ François Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
With the STUN server, I can ring the NATed phone!
But, I can hear the NATed phone with the public phone, and NATED phone can’t hear the public phone!
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : Asim Riaz [mailto:ariaz78@gmail.com] Envoyé : vendredi 20 février 2009 10:55 À : BERGANZ François Cc : Daniel-Constantin Mierla; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
On Fri, Feb 20, 2009 at 9:53 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:48 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:40 AM, BERGANZ François francois@acropolistelecom.net wrote:
I will try to install my stun server.
If I install my stun server, when I will do save("location"), it will save the good IP and port in the db?
yes, because UAs will know their public IPs using STUN.
you can also use NATHELPER Module in KAMAILIO to fix NAT e.g
if(nat_uac_test("1")) { fix_nated_register();
fix_nated_contact();
force_rport();
}
Thanks,
Asim
Cordialement, BERGANZ François
http://www.acropolistelecom.net
Pensez à lP'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 20 février 2009 10:22 À : BERGANZ François Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
_______________________________________________ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
With the STUN server, I can ring the NATed phone!
But, I can hear the NATed phone with the public phone, and NATED phone can’t hear the public phone!
àI just do
lookup("location") and t_relay()
is the problem comes from here?
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : Asim Riaz [mailto:ariaz78@gmail.com] Envoyé : vendredi 20 février 2009 10:55 À : BERGANZ François Cc : Daniel-Constantin Mierla; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
On Fri, Feb 20, 2009 at 9:53 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:48 AM, Asim Riaz ariaz78@gmail.com wrote:
On Fri, Feb 20, 2009 at 9:40 AM, BERGANZ François francois@acropolistelecom.net wrote:
I will try to install my stun server.
If I install my stun server, when I will do save("location"), it will save the good IP and port in the db?
yes, because UAs will know their public IPs using STUN.
you can also use NATHELPER Module in KAMAILIO to fix NAT e.g
if(nat_uac_test("1")) { fix_nated_register();
fix_nated_contact();
force_rport();
}
Thanks,
Asim
Cordialement, BERGANZ François
http://www.acropolistelecom.net
Pensez à lP'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 20 février 2009 10:22 À : BERGANZ François Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
Hello,
On 02/20/2009 10:03 AM, BERGANZ François wrote:
Hello,
Is it possible to do NAT traversal without RTP-PROXY… ?
If yes, have you an example?
if the phones are not behind symmetric NAT and they have STUN support, then you have to install a stun server or use a public one and nothing else to do on your sip server. It is so called client-side NAT traversal.
Cheers, Daniel
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
_______________________________________________ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
I wouldn't use STUN. It's an unnecessary complicated science project.
Unless the two media endpoints can't reach other, or the signaling agents can't reach other due to asymmetric signaling or bad NAT devices or stupidly implemented SIP ALGs or whatever, "nathelper" alone will do just fine. Just apply its fixups to your Contact URIs and SDP endpoints and get perfectly good far-end NAT traversal.
I have deployed this many times for service providers and it works great. There will always be problems with some flaky end-user equipment, but in general the benefits far outweigh the costs of screwing around with STUN or anything of the sort.
Have you an example?
Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Alex Balashov [mailto:abalashov@evaristesys.com] Envoyé : vendredi 20 février 2009 11:24 À : Asim Riaz Cc : BERGANZ François; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
I wouldn't use STUN. It's an unnecessary complicated science project.
Unless the two media endpoints can't reach other, or the signaling agents can't reach other due to asymmetric signaling or bad NAT devices or stupidly implemented SIP ALGs or whatever, "nathelper" alone will do just fine. Just apply its fixups to your Contact URIs and SDP endpoints and get perfectly good far-end NAT traversal.
I have deployed this many times for service providers and it works great. There will always be problems with some flaky end-user equipment, but in general the benefits far outweigh the costs of screwing around with STUN or anything of the sort.
The "nathelper" module documentation page really says everything there is to say on the subject.
BERGANZ François wrote:
Have you an example?
Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Alex Balashov [mailto:abalashov@evaristesys.com] Envoyé : vendredi 20 février 2009 11:24 À : Asim Riaz Cc : BERGANZ François; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
I wouldn't use STUN. It's an unnecessary complicated science project.
Unless the two media endpoints can't reach other, or the signaling agents can't reach other due to asymmetric signaling or bad NAT devices or stupidly implemented SIP ALGs or whatever, "nathelper" alone will do just fine. Just apply its fixups to your Contact URIs and SDP endpoints and get perfectly good far-end NAT traversal.
I have deployed this many times for service providers and it works great. There will always be problems with some flaky end-user equipment, but in general the benefits far outweigh the costs of screwing around with STUN or anything of the sort.
# nathelper
modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "natping_processes", 3) modparam("nathelper", "sipping_bflag", 3) modparam("nathelper", "sipping_from", "sip:pinger@place") modparam("nathelper", "sipping_method", "OPTIONS")
On initial INVITE received from user:
route[2} { if(nat_uac_test("1")) { fix_nated_contact(); force_rport(); }
if(search("Content-Type: application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); }
t_on_reply("1");
if(!t_relay()) { sl_reply_error(); exit; } }
## # Reply route for INVITE-related feedback and NAT fixups.
onreply_route[1] { # Technically, an SDP payload may be returned in # any non-100 1xx provisional message per RFC 3261, # not necessarily 180|183. So, decide how standards- # compliant you want to be.
if(t_check_status("200|180|183")) {
if(nat_uac_test("1")) fix_nated_contact();
if(search("Content-Type: application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } } }
On registration from user:
route[1} { if(nat_uac_test("1")) { fix_nated_contact(); force_rport(); setbflag(3); }
save("location");
}
BERGANZ François wrote:
Have you an example?
Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Alex Balashov [mailto:abalashov@evaristesys.com] Envoyé : vendredi 20 février 2009 11:24 À : Asim Riaz Cc : BERGANZ François; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
I wouldn't use STUN. It's an unnecessary complicated science project.
Unless the two media endpoints can't reach other, or the signaling agents can't reach other due to asymmetric signaling or bad NAT devices or stupidly implemented SIP ALGs or whatever, "nathelper" alone will do just fine. Just apply its fixups to your Contact URIs and SDP endpoints and get perfectly good far-end NAT traversal.
I have deployed this many times for service providers and it works great. There will always be problems with some flaky end-user equipment, but in general the benefits far outweigh the costs of screwing around with STUN or anything of the sort.
I tryed it, but I have that error:
ERROR:nathelper:mod_init: bad config - ping_nated_only enabled, but no nat bflag set in usrloc module ERROR:core:init_mod: failed to initialize module nathelper ERROR:core:main: error while initializing modules
I have: loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "natping_processes", 3) modparam("nathelper", "sipping_bflag", 3) modparam("nathelper", "sipping_from", "sip:555@217.x.x.x") modparam("nathelper", "sipping_method", "OPTIONS")
Cordialement, BERGANZ François
http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Alex Balashov [mailto:abalashov@evaristesys.com] Envoyé : vendredi 20 février 2009 11:47 À : BERGANZ François Cc : 'Asim Riaz'; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
# nathelper
modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "natping_processes", 3) modparam("nathelper", "sipping_bflag", 3) modparam("nathelper", "sipping_from", "sip:pinger@place") modparam("nathelper", "sipping_method", "OPTIONS")
On initial INVITE received from user:
route[2} { if(nat_uac_test("1")) { fix_nated_contact(); force_rport(); }
if(search("Content-Type: application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); }
t_on_reply("1");
if(!t_relay()) { sl_reply_error(); exit; } }
## # Reply route for INVITE-related feedback and NAT fixups.
onreply_route[1] { # Technically, an SDP payload may be returned in # any non-100 1xx provisional message per RFC 3261, # not necessarily 180|183. So, decide how standards- # compliant you want to be.
if(t_check_status("200|180|183")) {
if(nat_uac_test("1")) fix_nated_contact();
if(search("Content-Type: application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } } }
On registration from user:
route[1} { if(nat_uac_test("1")) { fix_nated_contact(); force_rport(); setbflag(3); }
save("location");
}
BERGANZ François wrote:
Have you an example?
Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Alex Balashov [mailto:abalashov@evaristesys.com] Envoyé : vendredi 20 février 2009 11:24 À : Asim Riaz Cc : BERGANZ François; users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
I wouldn't use STUN. It's an unnecessary complicated science project.
Unless the two media endpoints can't reach other, or the signaling agents can't reach other due to asymmetric signaling or bad NAT devices or stupidly implemented SIP ALGs or whatever, "nathelper" alone will do just fine. Just apply its fixups to your Contact URIs and SDP endpoints and get perfectly good far-end NAT traversal.
I have deployed this many times for service providers and it works great. There will always be problems with some flaky end-user equipment, but in general the benefits far outweigh the costs of screwing around with STUN or anything of the sort.
On Friday 20 February 2009, BERGANZ François wrote:
I tryed it, but I have that error:
ERROR:nathelper:mod_init: bad config - ping_nated_only enabled, but no nat bflag set in usrloc module ERROR:core:init_mod: failed to initialize module
Hi Francois,
just do what the error message suggest.. :-) You need to set the bflag parameter in the usrloc module too:
http://www.kamailio.org/docs/modules/devel/usrloc.html#id2451490
Cheers,
Henning
Ok, it is better. Now, I can call my phone NATed. But the RTP dont work in NATed area!
Have you any idea? Thank you
Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : Henning Westerholt [mailto:henning.westerholt@1und1.de] Envoyé : vendredi 20 février 2009 14:20 À : users@lists.kamailio.org Cc : BERGANZ François; 'Alex Balashov' Objet : Re: [Kamailio-Users] NAT traversal
On Friday 20 February 2009, BERGANZ François wrote:
I tryed it, but I have that error:
ERROR:nathelper:mod_init: bad config - ping_nated_only enabled, but no nat bflag set in usrloc module ERROR:core:init_mod: failed to initialize module
Hi Francois,
just do what the error message suggest.. :-) You need to set the bflag parameter in the usrloc module too:
http://www.kamailio.org/docs/modules/devel/usrloc.html#id2451490
Cheers,
Henning
2009/2/20 BERGANZ François francois@acropolistelecom.net:
Ok, it is better. Now, I can call my phone NATed. But the RTP dont work in NATed area!
Have you any idea?
Are you donig SIp captures and analyzing the address in SIP headers and SDP in order to debug the problem? or will you ask in the maillist for each new issue you get without checking it?
"phone A has sound but phone B doesn't have" has no easy explanation.
Hello,
I come back for NAT transversal. First, I am sorry if you think that I email a lot!
After some captures... I could see that the Nated phone tell in the 200ok that its RTP port is 192.168... "Peer audio RTP is at port 192.168.1.82:41000"
I just have to find why the phone say it. And how to open a RTP port and tell it to my asterisk trough the SER. Have you an idea?
Thank you
Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine----- De : users-bounces@lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] De la part de Iñaki Baz Castillo Envoyé : vendredi 20 février 2009 15:50 Cc : users@lists.kamailio.org Objet : Re: [Kamailio-Users] NAT traversal
2009/2/20 BERGANZ François francois@acropolistelecom.net:
Ok, it is better. Now, I can call my phone NATed. But the RTP dont work in NATed area!
Have you any idea?
Are you donig SIp captures and analyzing the address in SIP headers and SDP in order to debug the problem? or will you ask in the maillist for each new issue you get without checking it?
"phone A has sound but phone B doesn't have" has no easy explanation.
2009/2/23 BERGANZ François francois@acropolistelecom.net:
Hello,
I come back for NAT transversal. First, I am sorry if you think that I email a lot!
Well, you should understand that VoIp is complex, even more hen dealing with NAT, so there is not a magic solution and some knowledge is required.
After some captures... I could see that the Nated phone tell in the 200ok that its RTP port is 192.168... "Peer audio RTP is at port 192.168.1.82:41000"
I just have to find why the phone say it. And how to open a RTP port and tell it to my asterisk trough the SER. Have you an idea?
Please describe your topology: where is Asterisk? where are phones? NAT?
If Asterisk has public IP and phones are behind NAT you need a solution for the RTP which could be:
a) Using Asterisk comedia mode (so Asterisk ignores the private address in the phone's SDP and sends audio to the public address from which receives audio from the phone. This is achieved by setting "nat=yes" in the phone peer configuration.
b) Using a media proxy (RtpProxy or MediaProxy).
c) Using STUN in the phone (it requieres you have no symmetric NAT in your router). With it, the phone discovers which public address and port to write in the SIP headers and SDP so the other endpoint (Asterisk in your case) will see it as not natted.
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Alex Balashov -
Asim Riaz -
BERGANZ François -
Daniel-Constantin Mierla -
Henning Westerholt -
Iñaki Baz Castillo -
Mark Sayer