Re: [SR-Users] using kamailio to forward ipv6 to ipv4

> On Jan 5, 2017, at 12:24 PM, anfecora wrote: > > Hi guys hope you have a great new year. > I would appreciate if anyone can point me in to the right direction . > > I need to build a proxy to translate from ipv6 to ipv4, but kamailio should > not process the registers or

Re: [SR-Users] Kamailio behind ELB AWS

> On Jun 24, 2016, at 4:53 AM, yann christophe > wrote: > > Hello Everyone, > > I would like to know, if i can use ELB and route53 behind some > Kamailio servers for the high availibility ? It really depends on what you are trying to do. An ELB doesn’t help with

Re: [SR-Users] Rtpengine/rtpproxy and zrtp

Zrtp passes through rtpengine just fine. --FC Sent from my 6 plus On Aug 6, 2015, at 14:12, Alexandru Covalschi 568...@gmail.com wrote: Sorry if writing to wrong mailing list, I am very limited to traffic now amd don't know if there is any for rtpproxy/rtpengine. My question is - can

Re: [SR-Users] Re-invites from carrier breaks the call

Using a B2BUA would possibly be a solution. --FC Sent from my 6 plus On Feb 19, 2015, at 05:12, Alex Balashov abalas...@evaristesys.com wrote: ‎Hi Will, Unfortunately, there's not a clever workaround at your disposal here, of all scenarios. The SDP payload in the 200 OK must mimic that

Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)

On Feb 16, 2015, at 7:27 PM, Ovidiu Sas o...@voipembedded.com wrote: You could simply let the RTP traffic to flow directly between FS and endpoints (no need for rtpproxy). All you need to do is: - forward the appropriate RTP ports to FS; - fix the private IP in SDP by replacing it with the

Re: [SR-Users] R: Re: R: Re: RTPPROXY BRANCH

On Sep 29, 2014, at 1:14 PM, Richard Fuchs rfu...@sipwise.com wrote: This may work with rtpengine, as it will open new ports for answers come from different endpoints. But the final two-way association for the actual call may still end up broken, as it has no way of knowing which client

Re: [SR-Users] R: Re: R: Re: RTPPROXY BRANCH

On Sep 29, 2014, at 1:24 PM, Richard Fuchs rfu...@sipwise.com wrote: On 09/29/14 13:19, Frank Carmickle wrote: On Sep 29, 2014, at 1:14 PM, Richard Fuchs rfu...@sipwise.com wrote: This may work with rtpengine, as it will open new ports for answers come from different endpoints

Re: [SR-Users] R: Re: RTPPROXY BRANCH

On Sep 25, 2014, at 10:09 AM, Marino Mileti marino.mil...@alice.it wrote: Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive

Re: [SR-Users] R: Re: R: Re: RTPPROXY BRANCH

On Sep 25, 2014, at 10:41 AM, Marino Mileti marino.mil...@alice.it wrote: No no. The video will be sent by the caller user to all the callees. I'l try to explain better. My scenario is: - A make a call to a group... B C are group member...so Kamailio is able to call them in

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

On Sep 15, 2014, at 1:32 PM, Richard Fuchs rfu...@sipwise.com wrote: On 08/25/14 19:25, Alex Villací­s Lasso wrote: I have a rtpproxy configuration that spawns several rtpproxy instances, using bridge mode. An example is shown below: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u

Re: [SR-Users] Minimal configuration without database support

On Aug 27, 2014, at 5:48 AM, Eugene Prokopiev e...@itx.ru wrote: Have SEMS any advantages over FreeSWITCH for topology hiding? More narrow and specialised use-cases, so you don't have to deal with stripping it of PBX and miscellaneous application features. It's more of a dedicated B2BUA

Re: [SR-Users] Minimal configuration without database support

On Aug 26, 2014, at 9:46 AM, Eugene Prokopiev e...@itx.ru wrote: Is it possible at all to do double topology hiding with Kamailio? SIP devices with public ip addresses must not know private softswitch address and softswitch must not know any public ip address. Softswitch must work only with

Re: [SR-Users] rtpengine symmetric RTP behavior

On Aug 11, 2014, at 5:04 PM, Narsay, Deep deep.nar...@mezocliq.com wrote: Hello Andreas, Yes, that's what I had thought, but I am actually seeing it using two different UDP ports towards Freeswitch (send=40036 and recv=40042) and two more ports towards SIP Client (send=40038, and

Re: [SR-Users] Some of the calls drops after 15 minutes + some seconds

On Jul 16, 2014, at 4:05 PM, Andras FOGARASI fogar...@fogarasi.com wrote: On 7/16/14, 10:00 PM, Frank Carmickle wrote: On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I expect that the signaling is ok at least for call setup. From signling

Re: [SR-Users] Some of the calls drops after 15 minutes + some seconds

On Jul 17, 2014, at 11:27 AM, Andras FOGARASI fogar...@fogarasi.com wrote: On 7/17/14, 3:41 PM, Frank Carmickle wrote: I would expect that if it was a NAT issue you would see it much sooner than 15 minutes, 30-60 seconds. Are session timers being stripped by Kamailio? You say it's

Re: [SR-Users] Some of the calls drops after 15 minutes + some seconds

On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I expect that the signaling is ok at least for call setup. From signling point of view, I can think of following situations: - endpoints send keep alive packets (or session updates) which are no

Re: [SR-Users] TLS and SIP

On May 22, 2014, at 6:46 PM, James Cloos cloos+openser-us...@jhcloos.com wrote: If you record the full packet trace, wireshark can use your privkey.pem to decode the tls handshake, recover the session key, and use that to decode the payload packets. Cf http://wiki.wireshark.org/SSL for

Re: [SR-Users] Modifying SDP in Kamailio

On May 18, 2014, at 2:34 AM, MrIhaveAnOpinionOnEverything melry...@gmail.com wrote: Hi guys: I am a RD engineer trying to learn kamailio. After following some tutorials and reading the thread in this mailing list I was able to setup a voip backend with this configuration

Re: [SR-Users] TLS and SIP

On May 23, 2014, at 12:43 PM, James Cloos cl...@jhcloos.com wrote: FC == Frank Carmickle fr...@carmickle.com writes: JC If you record the full packet trace, wireshark can use your privkey.pem JC to decode the tls handshake, recover the session key, and use that to JC decode the payload

Re: [SR-Users] Forward copy of Register information to FreeSWITCH

On Mar 31, 2014, at 7:22 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: On 31/03/14 13:07, Alexandr Usov wrote: 2014-03-28 18:16 GMT+02:00 Frank Carmickle fr...@carmickle.com: Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail

Re: [SR-Users] Forward copy of Register information to FreeSWITCH

On Mar 28, 2014, at 11:36 AM, Alexandr Usov blessen...@gmail.com wrote: I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static host=kamailio_ip settings for each user on Asterisk side.

Re: [SR-Users] Kamailio Freeswitch

On Mar 22, 2014, at 3:18 AM, MrIhaveAnOpinionOnEverything melry...@gmail.com wrote: To whom it may concern: We are configuring a SIP platform with Kamailio and Freeswitch with this setup: UAC 1 == Kamailio == Freeswitch UAC 2 == Kamailio == Freeswitch We followed the

Re: [SR-Users] New official Debian and Ubuntu repository

On Mar 3, 2014, at 8:16 AM, Victor Seva linuxman...@torreviejawireless.org wrote: The new build system for Debian and Ubuntu packages is now in place. This service is kindly sponsored by SipWise [0] thanks to Andreas Granig [1]. Sipwise is providing the hosting and man power to create and

Re: [SR-Users] RTPProxy/Mediaproxy issue

On Feb 21, 2014, at 4:24 AM, Daniel Grotti dgro...@sipwise.com wrote: Hi, it looks like your platform/network is introducing jitter in RTP packets. Mediaproxy/rtpproxy usual introduce a very low jitter but it has no impact to performances at all. It's hard to say, you should investigate