Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server

[Pinter, Gerd.]

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.pin...@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  Im Auftrag von 
Daniel-Constantin Mierla
Gesendet: Dienstag, 8. Mai 2018 09:02
An: Kamailio (SER) - Users Mailing List 
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.  
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  
Gesendet: Montag, 7. Mai 2018 20:56
An: mailto:sr-users@lists.kamailio.org
Cc: Pinter, Gerd. 
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
 
 
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-- 
Dipl. Ing.
Markus Monka - mailto:mmo...@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



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Re: [SR-Users] kamailio + rtpproxy behind NAT

[Bonjour Madame]

Thank you for such a quick response. I will definitely give that article a
try.

Sincerely,
Clarence

On Tue, May 8, 2018, 6:15 PM Alex Balashov 
wrote:

> Hi Clarence,
>
> This article may help (in fairness, I wrote it):
>
>
> http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/
>
> On Tue, May 08, 2018 at 11:01:47PM +, Bonjour Madame wrote:
>
> > Hello,
> >
> > I have kamailio and rtpproxy running on the same server located behind a
> > router with NAT just like below.
> >
> > UA(internet)---public ip---router(NAT)--Kamailio+RTPproxy(private
> > ip)--UA(private network)
> >
> > There are several issues like one way audio.
> >
> > I need users on the internet to communicate with each other, users on the
> > private network to communicate with each other, and users on the internet
> > and the private network to communicate with each other.
> >
> > What is the best approach?
> >
> > Best regards,
> > Clarence
>
> > ___
> > Kamailio (SER) - Users Mailing List
> > sr-users@lists.kamailio.org
> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] kamailio + rtpproxy behind NAT

[Alex Balashov]

Hi Clarence,

This article may help (in fairness, I wrote it):

http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/

On Tue, May 08, 2018 at 11:01:47PM +, Bonjour Madame wrote:

> Hello,
> 
> I have kamailio and rtpproxy running on the same server located behind a
> router with NAT just like below.
> 
> UA(internet)---public ip---router(NAT)--Kamailio+RTPproxy(private
> ip)--UA(private network)
> 
> There are several issues like one way audio.
> 
> I need users on the internet to communicate with each other, users on the
> private network to communicate with each other, and users on the internet
> and the private network to communicate with each other.
> 
> What is the best approach?
> 
> Best regards,
> Clarence

> ___
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users


-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[SR-Users] Copy $xavp

[Duarte Rocha]

Greetings,

I'm getting a query result in a $xavp. I'm trying to copy the entire
structure to a new $xavp like this : $xavp(Valid_Acd_Results) =
$xavp(ACD_Query_Result[$var(iterator)]);

However, i can't retrieve the value with : xnotice(" Mode =
$xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing
something wrong?

Thanks
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[SR-Users] Kamailio + RTPEngine + RTPEngine Recording daemon forward-to option

[Balázs Radák]

Dear all,

I would need some help setting up Kamailio  with RTP engine.
I'm trying to make use of  RTPEngine recording daemon's forward-to option,
to have the calls streams sent to 3rd party application.

Have someone managed to have this forward-to option work?
If I understood well, when RTPEngine's forward-to option is turned on, the
call metainfo + streams can be sent to a socket server?


Thanks in advance!
Regards,
Balazs
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Re: [SR-Users] Record-route and proxy_challenge

[Евгений Голей]

Thanks for the answer! I did so yesterday, but I do not understand why when 
using functions this does not happen, plus is there a variable in which all the 
Record-Route headers are saved from the query?


>Понедельник,  7 мая 2018, 22:42 +03:00 от Alex Balashov 
>:
>
>None of that should be the case. Record-Route should be added to initial 
>invites only. And then it's the UASs job to copy the RR header into 
>dialog-forming replies (e.g. 2xx). 
>
>-- Alex
>
>--
>Sent via mobile, please forgive typos and brevity.


-- 
Evgeniy
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Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server

[Daniel-Constantin Mierla]

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.  > 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 >
> Gesendet: Montag, 7. Mai 2018 20:56
> An: sr-users@lists.kamailio.org 
> Cc: Pinter, Gerd. >
> 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
> sr-users@lists.kamailio.org 
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> 
>
>
>
>
> -- 
> Dipl. Ing.
> Markus Monka - mmo...@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@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

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Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server

[Daniel-Constantin Mierla]

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  
> Gesendet: Montag, 7. Mai 2018 20:56
> An: sr-users@lists.kamailio.org
> Cc: Pinter, Gerd. 
> 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
> sr-users@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



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Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server

[Markus Monka]

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.  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 
> Gesendet: Montag, 7. Mai 2018 20:56
> An: sr-users@lists.kamailio.org
> Cc: Pinter, Gerd. 
> 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
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Dipl. Ing.
Markus Monka - mmo...@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
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Re: [SR-Users] Garbage in to field

[Daniel-Constantin Mierla]

Hello,

this is somehow by design from early days of the application (2001-2003)
for speed purposes (at that time such optimizations matter for high
capacity) and it got propagated over the time.

More about it can be found in FAQ (I extended right now with a bit more
details about this specific case):

  -
https://www.kamailio.org/wiki/tutorials/faq/main#how_to_set_different_header_va

Cheers,
Daniel

On 08.05.18 01:11, Kjeld Flarup wrote:
> I'm having problems with my To field.
>
> I'm doing an async_route, and afterwards sends the call with auth to a
> SIP server.
>
> My problem is that in the first invite I see the expected To field
> sip:004520202...@isp.com
>
> But when the Auto is send the To field is changed to the, To field for
> the PBX
> sip:1234@pbx.local
>
>
> I tried to fix this with uac_replace_to("$ru"), but that ends up in
> this To field:
> sip:004520202020@isp.comsip:004520202...@isp.com
>
> It seems like uac_replace_to sometimes appends rather that replaces.
> If I run it twice, then it for sure appends. Is this expected behaviour.
>
>
>

-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com


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Re: [SR-Users] Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server

[Pinter, Gerd.]

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  
Gesendet: Montag, 7. Mai 2018 20:56
An: sr-users@lists.kamailio.org
Cc: Pinter, Gerd. 
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
 
 

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