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

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 (forme

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

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 wit

Re: [SR-Users] Garbage in to field

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 ca

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

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 comme

[SR-Users] How to change Calling Line ID in cr_route

I have cr_route working well. Only problem is that the username is sent off to the carrier and the 3 or 4 digit username appears as the callerid. Is there a module or scripts that can replace the username with a db table defined callerid. Say 202 maps to a 10 digit TN. KD  ___

Re: [SR-Users] STUN

The STUN solutions for NAT traversal is classified as near-end.This solutions implemented on the client-side, Then you don't need to configure kamailio. Be notice only the TURN solution is used on server-side. The TURN solution needs some configurations in Kamailio. With Regards.Mojtaba On Mon, M

Re: [SR-Users] STUN

No; STUN, by its nature, is a client-side solution. On Mon, May 07, 2018 at 12:59:49PM +, Wilkins, Steve wrote: > So If I already have a Stun Server, I don’t need to configure Kamailio to use > that Server…Is that correct? > > Thank you > > From: sr-users [mailto:sr-users-boun...@lists.kam

Re: [SR-Users] STUN

So If I already have a Stun Server, I don’t need to configure Kamailio to use that Server…Is that correct? Thank you From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Sergiu Pojoga Sent: Sunday, May 6, 2018 11:49 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [S

[SR-Users] Garbage in to field

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

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

On 07.05.18 17:34, Daniel Tryba wrote: > On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: >> Sure. Attached. Problem appears to be that the topos query can't find >> callid-totag (from the response). >> >> I'll try the same scenario with the mysql backend to see if it behaves >> diff

Re: [SR-Users] Record-route and proxy_challenge

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. ___ Kamail

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

[SR-Users] Record-route and proxy_challenge

Hi! Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions:   - proxy_challenge (),   - www_challenge (),   - send_reply (),   - sl_send_reply there is no Recor

Re: [SR-Users] Kamailio v5.1.3 Released

Looks as here is BSD Kamailio port https://www.freebsd.org/cgi/ports.cgi?query=kamailio Sergey пн, 7 мая 2018 г. в 13:09, Daniel-Constantin Mierla : > Hello, > > thanks for this! I guess we can remove the *BSD specs we have in > pkg/kamailio/, they are not maintained for very long. > > Does anyo

Re: [SR-Users] Determining next hop for any SIP message

Am Freitag, 4. Mai 2018, 16:25:06 CEST schrieb George Diamantopoulos: > Just one last attempt at clearing the following points, if someone feels > confident enough to answer: > > * Does $nh(d) work when the Route header enforces the next-hop in an > in-dialog request? > * Is it safe to determine n

Re: [SR-Users] Server-side NAT traversal with Kamailio

Well, that's not entirely true. Outbound requires significant server-side support. I just don't see it as adding much to NAT traversal, which works just fine without it. Outbound's value is really in solving a redundancy problem (through really Byzantine and bureaucratic means). On May 7, 201

Re: [SR-Users] Route header in local reply

Sorry, figured out, the question is not relevant. >Понедельник, 7 мая 2018, 20:29 +03:00 от Евгений Голей : > >Hi. > >Colleagues write how to configure the local responses to requests so that the >response added the header of the Route in accordance with the record-route >headers that were in

[SR-Users] Route header in local reply

Hi. Colleagues write how to configure the local responses to requests so that the response added the header of the Route in accordance with the record-route headers that were in the request ? Example SBC --> Kamailio REGISTER sip:domain.com SIP/2.0 Record-Route: Via: SIP/2.0/UDP CCC.CCC.CCC

Re: [SR-Users] Server-side NAT traversal with Kamailio

BTW, my quarrel with add_contact_alias() vs fix_nated_register() comes from the fact that when forwarding registration onward, Asterisk for example chooses to display the peer's host address as Kamailio's address instead of the device's NATed address or even worse the Private address, which isn't t

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos b

Re: [SR-Users] How long time does a push take?

Once you have sent the push, you are done... now wait for them to deliver it to your device. Fingers crossed. Apple VoIP push are good and take ~ 1 sec to deliver to the target device if there is no lag on internet leyer. Its the mobile app that take further time to wake up based on your defined cr

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

On Mon, May 07, 2018 at 12:36:29PM +0200, Daniel-Constantin Mierla wrote: > can you give the logs with debug=3 for the entire call (from initial > INVITE)? Sure. Attached. Problem appears to be that the topos query can't find callid-totag (from the response). I'll try the same scenario with the m

Re: [SR-Users] Server-side NAT traversal with Kamailio

Awesome.  Thanks KD On Monday, May 7, 2018, 7:27:37 AM EDT, Alex Balashov wrote: Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/ This is a rough cut, and may be edited or expand

Re: [SR-Users] Pleasing all kinds of UACs for RTP/SRTP with rtpengine

Hi Daniel, Thar is correct, the top Via header branch param differs (has a .1 instead of .0 at the end), that doesn't seem to count according to RFC 3271 section 8.2.2.2 Looking at the sip trace, the ACK for 415 is sent out before the 2nd bra

Re: [SR-Users] Server-side NAT traversal with Kamailio

On a second thought, SIP Outbound is client-side oriented, so never mind, I guess. On Mon, May 7, 2018 at 8:49 AM, Sergiu Pojoga wrote: > Hi Alex, > > Very thorough write up, as always. If only this article was available some > time ago when I was putting up an edge proxy, would have saved me we

Re: [SR-Users] Server-side NAT traversal with Kamailio

Hi Alex, Very thorough write up, as always. If only this article was available some time ago when I was putting up an edge proxy, would have saved me weeks of frustration in figuring out all these NAT-related aspects and RFC readings (although such an exercise helps understand the problem to its c

[SR-Users] Server-side NAT traversal with Kamailio

Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/ This is a rough cut, and may be edited or expanded later. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-80

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

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 t

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

Hello, can you give the logs with debug=3 for the entire call (from initial INVITE)? Cheers, Daniel On 04.05.18 16:34, Daniel Tryba wrote: > Enabling topos breaks PRACK in my simple (near default) proxy setup > (attached kamailio.conf). > > asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <->

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://git

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

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

Re: [SR-Users] Pleasing all kinds of UACs for RTP/SRTP with rtpengine

Hello, the branch parameter value in top Via header is different, so this should indicate is not the same branch. Have you look at the network traffic, is the ACK for 415 getting first to device, before the 2nd branch INVITE? A that moment, the transaction should be considered terminated and a ne

Re: [SR-Users] Kamailio v5.1.3 Released

Hello, thanks for this! I guess we can remove the *BSD specs we have in pkg/kamailio/, they are not maintained for very long. Does anyone know if there are corresponding ports on NetBSD or OpenBSD? Cheers, Daniel On 06.05.18 12:21, Ben Hood wrote: > Many thanks for this Daniel. > > FYI the off

Re: [SR-Users] STUN

Hi, Let me explain a bit the STUN server in kamailio is for SIP signaling to help resolve NAT issues related to SIP over UDP for SIP clients. While a separate / 3rd party STUN server is mostly useful for NAT issues related to media problems. Thank you. On Mon, 7 May 2018 at 10:33, Mojtaba wrote:

Re: [SR-Users] STUN

Hello, You could run STUN server on the same server that Kamailio is running, It is very straightforward. Just donload STUNTMAN ,An open source STUN server - Version 1.2.7, and read README file. Alternative, Opensips has build-in STUN module, You could use it,too With Regards.Mojtaba On Sun, May