7 May
2018
7 May
'18
3:22 p.m.
Hi there,
unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway, I
have to maintain a SIP Server for high quality Audio transmission with special UA's.
Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I
Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP
address issues" most providers use private ranges separated by CGNAT from the real
Internet. This causes a lot of trouble. In some cases stun server entries on the clients
helps, in other cases don't!
But these days I read in announcement of a german company (Mayah Communications), which is
in the ProAudio biz since decades, that they offer a SIP Service where no stun is
recommended whether the clients are behind NAT or not.
After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that
there might exist a solution for my Kamailio SIP server to make it work like the
"Mayah" Sip Server.
What settings have to be done that Kamailio uses IP Header Information for SIP signalling
instead of the content of SIP packages themselves. Without stun you find the Local Address
in the SIP packages, and with Stun the external Address but mostly with the wrong
portnumber, but in the IP header you'll find the correct IP:port.
I already read a lot of the Kamailio documentation but due to the fact that I am not too
deep in IT I am not sure if I got it all. If someone could give me a useful hint on that,
that would make my day.
Hope this description is unterstandable.
Nice Regards
Gerhard Pinter
7 May
7 May
3:33 p.m.
Hello,
Kamailio can help with NAT traversal -- there are couple of options, one
of the most common deployments is using nathelper and rtpengine (or
rtpproxy).
A similar solution is provided in the default kamailio.cfg, just look at
WITH_NAT ifdefs to figure out the related parts:
- https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
Cheers,
Daniel
On 07.05.18 12:22, Pinter, Gerd. wrote:
Hi there,
unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway,
I have to maintain a SIP Server for high quality Audio transmission with special
UA's.
Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I
Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP
address issues" most providers use private ranges separated by CGNAT from the real
Internet. This causes a lot of trouble. In some cases stun server entries on the clients
helps, in other cases don't!
But these days I read in announcement of a german company (Mayah Communications), which
is in the ProAudio biz since decades, that they offer a SIP Service where no stun is
recommended whether the clients are behind NAT or not.
After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that
there might exist a solution for my Kamailio SIP server to make it work like the
"Mayah" Sip Server.
What settings have to be done that Kamailio uses IP Header Information for SIP signalling
instead of the content of SIP packages themselves. Without stun you find the Local Address
in the SIP packages, and with Stun the external Address but mostly with the wrong
portnumber, but in the IP header you'll find the correct IP:port.
I already read a lot of the Kamailio documentation but due to the fact that I am not too
deep in IT I am not sure if I got it all. If someone could give me a useful hint on that,
that would make my day.
Hope this description is unterstandable.
Nice Regards
Gerhard Pinter
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
4:11 p.m.
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit for all IT traffic, and
our Sip Server also have to manage connections outbound our house, where I thought it
might be better let those clients do the payload by peer to peer connection. If I got it
right this traffic would flow thru our Sip server with RTP Proxy enabled.
Thanks a lot
Best Regards Gerd
-----Ursprüngliche Nachricht-----
Von: Daniel-Constantin Mierla <miconda(a)gmail.com>
Gesendet: Montag, 7. Mai 2018 12:34
An: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>rg>; Pinter, Gerd.
<G.Pinter(a)radionrw.de>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a
Stun Server
Hello,
Kamailio can help with NAT traversal -- there are couple of options, one of the most
common deployments is using nathelper and rtpengine (or rtpproxy).
A similar solution is provided in the default kamailio.cfg, just look at WITH_NAT ifdefs
to figure out the related parts:
- https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
Cheers,
Daniel
On 07.05.18 12:22, Pinter, Gerd. wrote:
Hi there,
unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway,
I have to maintain a SIP Server for high quality Audio transmission with special
UA's.
Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I
Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP
address issues" most providers use private ranges separated by CGNAT from the real
Internet. This causes a lot of trouble. In some cases stun server entries on the clients
helps, in other cases don't!
But these days I read in announcement of a german company (Mayah Communications), which
is in the ProAudio biz since decades, that they offer a SIP Service where no stun is
recommended whether the clients are behind NAT or not.
After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that
there might exist a solution for my Kamailio SIP server to make it work like the
"Mayah" Sip Server.
What settings have to be done that Kamailio uses IP Header Information for SIP signalling
instead of the content of SIP packages themselves. Without stun you find the Local Address
in the SIP packages, and with Stun the external Address but mostly with the wrong
portnumber, but in the IP header you'll find the correct IP:port.
I already read a lot of the Kamailio documentation but due to the fact that I am not too
deep in IT I am not sure if I got it all. If someone could give me a useful hint on that,
that would make my day.
Hope this description is unterstandable.
Nice Regards
Gerhard Pinter
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May
14-16, 2018 - www.kamailioworld.com
11:56 p.m.
New subject: Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server
Am Montag, 7. Mai 2018, 13:11:50 CEST schrieb Pinter, Gerd.:
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit for all IT
traffic, and our Sip Server also have to manage connections outbound our
house, where I thought it might be better let those clients do the payload
by peer to peer connection. If I got it right this traffic would flow thru
our Sip server with RTP Proxy enabled.
Thanks a lot
Hello Gerd,
you understood it correctly, indeed using rtpengine/rtpproxy would mean routing
additionally your RTP traffic to your network. Depending on the number of sessions and
the used codec it may work perfectly, only with some QoS tuning or not at all. You can
estimate the bandwith, there are also some calculators online.
Best regards,
Henning Westerholt
8 May
8 May
11:29 a.m.
Hello Henning, hello Community
we are using high quality Codecs with bitrates up to 1500Kbit (PCM), but usually 128kbit
(Mpeg1 Layer3 or AAC-LD). 128 is Not too much, but I want to be on the save side and
therefore I prefer solution without Proxy.
By the way I am audio engineer at a german commercial radio broadcast network and we try
to establish our own "ISDN replacement-SIP Network" for reporting from events
like Soccer Games, Karneval, Rock Concerts, Political Party Summits, Lawsuits...
whatever. I want to keep frontend as simple as it gets for the reporter.
Anyway, people told me that the SIP Server of "Mayah Communications" is working
without the need of stun or other Client side gadgets, but I am also told that this Mayah
server works without Proxy functions. That Company won't tell me their settings of
course, but if possible I'd like to have this feature for our SIP Server. At the
moment I have still vast problems with connectivity from Devices that are logged on via
Vodafone Mobile Network. Even Stun does not work. I guess that CGNAT of Vodafone (All of
the Devices I have tested within Vodafone allocate IP Addresses in private range!) opens
different Ports for different outbound connections which is the case if I use a 3rd party
Stun Server. I have read a lot about this issue these days and there is solution, but I am
not too deep into scripting an how to edit the kamailio.cfg. For example, I tried to start
Kamailio with Stun, but if I tried to use the built in Stun, I've got error messages
on the Client and also in the logfile of the SIP Server (incomplete header of Stun
message) Anyone can help, please?
Best regards Gerd
Von: Henning Westerholt <hw(a)kamailio.org>
Gesendet: Montag, 7. Mai 2018 20:56
An: sr-users(a)lists.kamailio.org
Cc: Pinter, Gerd. <G.Pinter(a)radionrw.de>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a
Stun Server
Am Montag, 7. Mai 2018, 13:11:50 CEST schrieb Pinter, Gerd.:
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit for all IT
traffic, and our Sip Server also have to manage connections outbound our
house, where I thought it might be better let those clients do the payload
by peer to peer connection. If I got it right this traffic would flow thru
our Sip server with RTP Proxy enabled.
Thanks a lot
Hello Gerd,
you understood it correctly, indeed using rtpengine/rtpproxy would mean routing
additionally your RTP traffic to your network. Depending on the number of sessions and the
used codec it may work perfectly, only with some QoS tuning or not at all. You can
estimate the bandwith, there are also some calculators online.
Best regards,
Henning Westerholt
11:40 a.m.
Hi Gerd,
have you seen Olle's Project from 2017 about RadioBroadcasting?
https://www.kamailio.org/w/2017/11/oss-iris-broadcast-project-launched/
Best Regards
Markus
On Tue, May 8, 2018 at 8:29 AM, Pinter, Gerd. <G.Pinter(a)radionrw.de> wrote:
Hello Henning, hello Community
we are using high quality Codecs with bitrates up to 1500Kbit (PCM), but
usually 128kbit (Mpeg1 Layer3 or AAC-LD). 128 is Not too much, but I want
to be on the save side and therefore I prefer solution without Proxy.
By the way I am audio engineer at a german commercial radio broadcast
network and we try to establish our own "ISDN replacement-SIP Network" for
reporting from events like Soccer Games, Karneval, Rock Concerts, Political
Party Summits, Lawsuits... whatever. I want to keep frontend as simple as
it gets for the reporter.
Anyway, people told me that the SIP Server of "Mayah Communications" is
working without the need of stun or other Client side gadgets, but I am
also told that this Mayah server works without Proxy functions. That
Company won't tell me their settings of course, but if possible I'd like to
have this feature for our SIP Server. At the moment I have still vast
problems with connectivity from Devices that are logged on via Vodafone
Mobile Network. Even Stun does not work. I guess that CGNAT of Vodafone
(All of the Devices I have tested within Vodafone allocate IP Addresses in
private range!) opens different Ports for different outbound connections
which is the case if I use a 3rd party Stun Server. I have read a lot about
this issue these days and there is solution, but I am not too deep into
scripting an how to edit the kamailio.cfg. For example, I tried to start
Kamailio with Stun, but if I tried to use the built in Stun, I've got error
messages on the Client and also in the logfile of the SIP Server
(incomplete header of Stun message) Anyone can help, please?
Best regards Gerd
Von: Henning Westerholt <hw(a)kamailio.org>
Gesendet: Montag, 7. Mai 2018 20:56
An: sr-users(a)lists.kamailio.org
Cc: Pinter, Gerd. <G.Pinter(a)radionrw.de>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio
without using a Stun Server
Am Montag, 7. Mai 2018, 13:11:50 CEST schrieb Pinter, Gerd.:
--
Dipl. Ing.
Markus Monka - mmonka(a)sipgate.de
Telefon: +49 (0)211-63 55 55-23
Telefax: +49 (0)211-63 55 55-22
sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit for all IT
traffic, and our Sip Server also have to manage connections outbound our
house, where I thought it might be better let those clients do the
payload
by peer to peer connection. If I got it right
this traffic would flow
thru
our Sip server with RTP Proxy enabled.
Thanks a lot
Hello Gerd,
you understood it correctly, indeed using rtpengine/rtpproxy would mean
routing additionally your RTP traffic to your network. Depending on the
number of sessions and the used codec it may work perfectly, only with some
QoS tuning or not at all. You can estimate the bandwith, there are also
some calculators online.
Best regards,
Henning Westerholt
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
12:02 p.m.
Hello,
just to add the link to the video of the presentation on this topic done
by Olle at the Kamailio World Conference 2017:
- https://www.youtube.com/watch?v=50TFZWlhsP4
Cheers,
Daniel
On 08.05.18 08:40, Markus Monka wrote:
Hi Gerd,
have you seen Olle's Project from 2017 about RadioBroadcasting?
https://www.kamailio.org/w/2017/11/oss-iris-broadcast-project-launched/
Best Regards
Markus
On Tue, May 8, 2018 at 8:29 AM, Pinter, Gerd. <G.Pinter(a)radionrw.de
<mailto:G.Pinter@radionrw.de>> wrote:
Hello Henning, hello Community
we are using high quality Codecs with bitrates up to 1500Kbit
(PCM), but usually 128kbit (Mpeg1 Layer3 or AAC-LD). 128 is Not
too much, but I want to be on the save side and therefore I prefer
solution without Proxy.
By the way I am audio engineer at a german commercial radio
broadcast network and we try to establish our own "ISDN
replacement-SIP Network" for reporting from events like Soccer
Games, Karneval, Rock Concerts, Political Party Summits,
Lawsuits... whatever. I want to keep frontend as simple as it
gets for the reporter.
Anyway, people told me that the SIP Server of "Mayah
Communications" is working without the need of stun or other
Client side gadgets, but I am also told that this Mayah server
works without Proxy functions. That Company won't tell me their
settings of course, but if possible I'd like to have this feature
for our SIP Server. At the moment I have still vast problems with
connectivity from Devices that are logged on via Vodafone Mobile
Network. Even Stun does not work. I guess that CGNAT of Vodafone
(All of the Devices I have tested within Vodafone allocate IP
Addresses in private range!) opens different Ports for different
outbound connections which is the case if I use a 3rd party Stun
Server. I have read a lot about this issue these days and there is
solution, but I am not too deep into scripting an how to edit the
kamailio.cfg. For example, I tried to start Kamailio with Stun,
but if I tried to use the built in Stun, I've got error messages
on the Client and also in the logfile of the SIP Server
(incomplete header of Stun message) Anyone can help, please?
Best regards Gerd
Von: Henning Westerholt <hw(a)kamailio.org <mailto:hw@kamailio.org>>
Gesendet: Montag, 7. Mai 2018 20:56
An: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
Cc: Pinter, Gerd. <G.Pinter(a)radionrw.de <mailto:G.Pinter@radionrw.de>>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with
Kamailio without using a Stun Server
Am Montag, 7. Mai 2018, 13:11:50 CEST schrieb Pinter, Gerd.:
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit
for all IT
traffic, and our Sip Server also have to manage
connections
outbound our
house, where I thought it might be better let
those clients do
the payload
by peer to peer connection. If I got it right
this traffic would
flow thru
our Sip server with RTP Proxy enabled.
Thanks a lot
Hello Gerd,
you understood it correctly, indeed using rtpengine/rtpproxy would
mean routing additionally your RTP traffic to your network.
Depending on the number of sessions and the used codec it may work
perfectly, only with some QoS tuning or not at all. You can
estimate the bandwith, there are also some calculators online.
Best regards,
Henning Westerholt
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
Dipl. Ing.
Markus Monka - mmonka(a)sipgate.de <mailto:mmonka@sipgate.de>
Telefon: +49 (0)211-63 55 55-23
Telefax: +49 (0)211-63 55 55-22
sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
9 May
9 May
10:22 a.m.
Hello there,
thanks for all the good hints and links. Guidance here was very helpful. I configured our
Kamailio SIP as Rttpproxy and everything works fine as I supposed.
A special Applause to Alex Balashov for the Blog wich is very imformative on this issue.
*Ticket closed*
P.S.
Thanks for all the fish, and never forget the towel
With nice regards
Mit freundlichen Grüßen
Gerhard Pinter
Digital- Studiotechnik
mailto:g.pinter@radionrw.de
Tel: 0208-85 87-312
Fax: 0208-85 87-480
radio NRW GmbH
Essener Str. 55
46047 Oberhausen
SITZ OBERHAUSEN
Amtsgericht Duisburg
Nr. HRB 12489
GESCHÄFTSFÜHRER
Sven Thölen
AUFSICHTSRATSVORSITZENDER
Hans-Jürgen Weske
Von: sr-users <sr-users-bounces(a)lists.kamailio.org> Im Auftrag von Daniel-Constantin
Mierla
Gesendet: Dienstag, 8. Mai 2018 09:02
An: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a
Stun Server
Hello,
just to add the link to the video of the presentation on this topic done by Olle at the
Kamailio World Conference 2017:
- https://www.youtube.com/watch?v=50TFZWlhsP4
Cheers,
Daniel
On 08.05.18 08:40, Markus Monka wrote:
Hi Gerd,
have you seen Olle's Project from 2017 about RadioBroadcasting?
https://www.kamailio.org/w/2017/11/oss-iris-broadcast-project-launched/
Best Regards
Markus
On Tue, May 8, 2018 at 8:29 AM, Pinter, Gerd. <mailto:G.Pinter@radionrw.de> wrote:
Hello Henning, hello Community
we are using high quality Codecs with bitrates up to 1500Kbit (PCM), but usually 128kbit
(Mpeg1 Layer3 or AAC-LD). 128 is Not too much, but I want to be on the save side and
therefore I prefer solution without Proxy.
By the way I am audio engineer at a german commercial radio broadcast network and we try
to establish our own "ISDN replacement-SIP Network" for reporting from events
like Soccer Games, Karneval, Rock Concerts, Political Party Summits, Lawsuits...
whatever. I want to keep frontend as simple as it gets for the reporter.
Anyway, people told me that the SIP Server of "Mayah Communications" is working
without the need of stun or other Client side gadgets, but I am also told that this Mayah
server works without Proxy functions. That Company won't tell me their settings of
course, but if possible I'd like to have this feature for our SIP Server. At the
moment I have still vast problems with connectivity from Devices that are logged on via
Vodafone Mobile Network. Even Stun does not work. I guess that CGNAT of Vodafone (All of
the Devices I have tested within Vodafone allocate IP Addresses in private range!) opens
different Ports for different outbound connections which is the case if I use a 3rd party
Stun Server. I have read a lot about this issue these days and there is solution, but I am
not too deep into scripting an how to edit the kamailio.cfg. For example, I tried to start
Kamailio with Stun, but if I tried to use the built in Stun, I've got error messages
on the Client and also in the logfile of the SIP Server (incomplete header of Stun
message) Anyone can help, please?
Best regards Gerd
Von: Henning Westerholt <mailto:hw@kamailio.org>
Gesendet: Montag, 7. Mai 2018 20:56
An: mailto:sr-users@lists.kamailio.org
Cc: Pinter, Gerd. <mailto:G.Pinter@radionrw.de>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a
Stun Server
Am Montag, 7. Mai 2018, 13:11:50 CEST schrieb Pinter, Gerd.:
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit for all IT
traffic, and our Sip Server also have to manage connections outbound our
house, where I thought it might be better let those clients do the payload
by peer to peer connection. If I got it right this traffic would flow thru
our Sip server with RTP Proxy enabled.
Thanks a lot
Hello Gerd,
you understood it correctly, indeed using rtpengine/rtpproxy would mean routing
additionally your RTP traffic to your network. Depending on the number of sessions and the
used codec it may work perfectly, only with some QoS tuning or not at all. You can
estimate the bandwith, there are also some calculators online.
Best regards,
Henning Westerholt
_______________________________________________
Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Dipl. Ing.
Markus Monka - mailto:mmonka@sipgate.de
Telefon: +49 (0)211-63 55 55-23
Telefax: +49 (0)211-63 55 55-22
sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
_______________________________________________
Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.twitter.com/miconda -- http://www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - http://www.kamailioworld.com
8 May
8 May
11:54 a.m.
Hello,
there is no way of going through symmetric NAT without a media stream
relay on a public ip address.
The media relay system can be:
1) server side relay controlled by sip server, like rtpengine or rtpproxy
2) dedicated SBC boxes put between end device and sip server, like
Oracle (former ACME) SBC
3) TURN servers, which is concept similar to rtpengine/rtpproxy, but
it is the end device interacting directly with it (sip server does not
interact with the TURN server), so the end device must support TURN protocol
If you do not control the end device or the end device doesn't support
TURN, then the solution has to be 1) or 2). The 2) can be expensive and
adds additional ongoing operations costs.
Note that you can put rtpproxy/rtpengine on a different system than
kamailio. Also, there can be many of them, with kamailio doing sort of
load balancing to distribute calls across all available
rtpproxy/rtpengine instance.
The is an alternative by creating a VPN between end devices and core
infrastructure, so everyone is in the network. However, all packets,
including the RTP/media streams are relayed by the VPN server, so you
still get the traffic in the core network.
Cheers,
Daniel
On 08.05.18 08:29, Pinter, Gerd. wrote:
Hello Henning, hello Community
we are using high quality Codecs with bitrates up to 1500Kbit (PCM), but usually 128kbit
(Mpeg1 Layer3 or AAC-LD). 128 is Not too much, but I want to be on the save side and
therefore I prefer solution without Proxy.
By the way I am audio engineer at a german commercial radio broadcast network and we try
to establish our own "ISDN replacement-SIP Network" for reporting from events
like Soccer Games, Karneval, Rock Concerts, Political Party Summits, Lawsuits...
whatever. I want to keep frontend as simple as it gets for the reporter.
Anyway, people told me that the SIP Server of "Mayah Communications" is working
without the need of stun or other Client side gadgets, but I am also told that this Mayah
server works without Proxy functions. That Company won't tell me their settings of
course, but if possible I'd like to have this feature for our SIP Server. At the
moment I have still vast problems with connectivity from Devices that are logged on via
Vodafone Mobile Network. Even Stun does not work. I guess that CGNAT of Vodafone (All of
the Devices I have tested within Vodafone allocate IP Addresses in private range!) opens
different Ports for different outbound connections which is the case if I use a 3rd party
Stun Server. I have read a lot about this issue these days and there is solution, but I am
not too deep into scripting an how to edit the kamailio.cfg. For example, I tried to start
Kamailio with Stun, but if I tried to use the built in Stun, I've got error messages
on the Client and also in the logfile of the SIP Server (incomplete header of Stun
message) Anyone can help, please?
Best regards Gerd
Von: Henning Westerholt <hw(a)kamailio.org>
Gesendet: Montag, 7. Mai 2018 20:56
An: sr-users(a)lists.kamailio.org
Cc: Pinter, Gerd. <G.Pinter(a)radionrw.de>
Betreff: Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using
a Stun Server
Am Montag, 7. Mai 2018, 13:11:50 CEST schrieb Pinter, Gerd.:
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Hi
Won't RTP Proxy cause al lot of traffic? We only have 155mbit for all IT
traffic, and our Sip Server also have to manage connections outbound our
house, where I thought it might be better let those clients do the payload
by peer to peer connection. If I got it right this traffic would flow thru
our Sip server with RTP Proxy enabled.
Thanks a lot
Hello Gerd,
you understood it correctly, indeed using rtpengine/rtpproxy would mean routing
additionally your RTP traffic to your network. Depending on the number of sessions and the
used codec it may work perfectly, only with some QoS tuning or not at all. You can
estimate the bandwith, there are also some calculators online.
Best regards,
Henning Westerholt
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8 comments
4 participants
participants (4)
-
Daniel-Constantin Mierla -
Henning Westerholt -
Markus Monka -
Pinter, Gerd.