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 invites, only translate from ipv6 to ipv4 and > forward all registrations over. A bit more information about the situation would be helpful. What exactly are you trying to do? > is that possible? 6 to 4 translation is possible, but until we understand the situation I’m not sure we can be that helpful. Are these phones? If so are they dual stack? Is the underlying network dual stack? There’s a lot of work to do to get this right in a Kamailio config. You’ll need to run rtpengine to proxy the media if the endpoints are not dual stack. Who keeps the mapping of the v6 to v4 and v4 to v6 translations? How is that mapping determined? I’d recommend ds-lite if you are in a position to alter the network and the phones/systems allow for it. HTH —FC > thanks. > > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 SIP routing. > I use Kamailio behind Freeswitch. I would like deploy these servers on AWS. > > What do you suggest me for the high availibility on AWS with Kamailio. SRV works well for most people. —FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 they support ZRTP at least in pass-through mode? Will > rtpengine fail on trying to recognize unknown SDP fields? > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 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 endpoint's previous > SDP answer in order for there to be media continuity. There's just a whole > lot of state you can't keep spoof in Kamailio. > > No, alas, it's one of those cases where basic SIP compliance issues in the > endpoints really must be fixed. Sorry. > > -- > Sent from my BlackBerry. Please excuse errors and brevity. > From: Will Ferrer > Sent: Wednesday, February 18, 2015 9:44 PM > To: Kamailio (SER) - Users Mailing List > Reply To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Re-invites from carrier breaks the call > > Hi Alex > > Thanks so much for the reply. > > Is there anything that we could do perhaps that is a more creative solution, > for instance not passing the re-invite all the way to the softphone and just > responding from the kamailio box handling the call? > > We tried this as well actually, but we didn't get it to work. We just sent a > 200 ok from the kamailio box, no sdp or anything on the packet since we sent > it with just send_reply and the carrier just sent a bye. > > Hopefully there is something clever we could do to correct the problem, it is > preventing us from using alot of our carriers since the re-invite breaks our > clients softphones. > > Thanks again for the assistance. > > All the best. > > Will Ferrer > >> On Wed, Feb 18, 2015 at 6:07 PM, Alex Balashov >> wrote: >> Kamailio cannot correct this. This is an endpoint issue. The whole point of >> Record-Route is to hairpin sequential requests (and indeed, their replies) >> through the proxy. The endpoints need to comply by affixing the correct >> Route header to the end-to-end ACK. >> >> -- >> Sent from my BlackBerry. Please excuse errors and brevity. >> From: Will Ferrer >> Sent: Wednesday, February 18, 2015 9:01 PM >> To: Kamailio (SER) - Users Mailing List >> Reply To: Kamailio (SER) - Users Mailing List >> Subject: [SR-Users] Re-invites from carrier breaks the call >> >> Hi All >> >> We have any issue with re invites coming from the carrier. >> >> When a reinvite occurs, our softphone client gets the invite, sends a 100, >> and then sends 200 ok. However the 200 ok does not have the softphones ip in >> the record route. Since it's not in the record route the ack from the >> carrier never makes it's way all the back to the softphone. >> >> This causes the softphone to keep sending 200 oks since it never gets the >> ack. >> >> Eventually the softphone gets tired of sending 200 oks and sends a bye. >> >> Is there any way that Kamailio can help me correct for this, or do we need >> to have our clients use different softphones? If it has to be handled via >> softphones is there even a softphone that can account for this? >> >> Thanks for all your assistance in advance. >> >> All the best. >> >> Will Ferrer >> >> Switchsoft >> >> >> >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge "ie" "ei" behind NAT (like in aws EC2)
On Feb 16, 2015, at 7:27 PM, Ovidiu Sas 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 public IP for > the inbound rtp streams (to FS). > FS will do this for you if you set the ext-rtp-ip on the profile. —FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] async workers
On Oct 24, 2014, at 9:15 AM, Daniel-Constantin Mierla wrote: > > On 23/10/14 04:03, Alex Balashov wrote: >> Also, what is the point of core async_workers setting versus the >> 'workers' modparam to async? Are they supposed to equal each other? >> Does one override the other? > > async_workers from core are common for all modules, being a decision not > to have each module that wants async operations to create its own pool > of processes. The workers defined by async module are only for that > module and used only by async_route() and async_sleep(). > > The implementation is also different, the async module workers are more > like timer processes (because both of async_route() and async_sleep() > need to sleep some interval of time). The module itself keeps the lists > of tasks in a structure optimized for timer execution. Each of this > async module workers check from time to time to see if there is a task > to be executed, executes what matches the time, then sleeps again for > 100ms (iirc), then checks again... > > The async_workers from core were designed to receive the job > immediately. Because of that, there is an interprocess communication > based on sockets in memory. The async workers are listening on them, so > once a sip worker sends the task to them, an async worker will receive it. I don't understand this stuff at all. I do know that when Freeswitch started using timerfd these sorts of issues got better by quite a bit. Maybe this would help here? Maybe you're already using this? --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Registrar module and local paths
On Oct 15, 2014, at 10:40 AM, Ben Langfeld wrote: > On 15 October 2014 11:32, Juha Heinanen wrote: > Ben Langfeld writes: > > > I figure at this point it may be simpler to separate the registrar and the > > proxy rather than attempt to debug this further, though if you have any > > other suggestions to avoid that I'd love to hear them. > > one possibility is that both of your combined proxy/registrars have > their own location tables and you forward registrations from one to the > other. > > The problem with that is horizontal scalability brings noise. If I have 10 of > these things, the SIP replication alone would be flooding the network. If you are running in an environment where you can use multicast it might be an option for you. Multicast the registrations from the edge proxy to the registrar cluster. If not maybe you can get the registrars to replicate to each other on a separate interface from the interface facing the edge proxy. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
On Sep 29, 2014, at 1:24 PM, Richard Fuchs wrote: > On 09/29/14 13:19, Frank Carmickle wrote: >> >> On Sep 29, 2014, at 1:14 PM, Richard Fuchs 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 ends up receiving the call (unless they do a final re-invite). >> >> But it should see the 200. Shouldn't that be enough? > > Unfortunately it doesn't see SIP codes, it only sees SDP bodies and some > particular details from the SIP message. 200 OK would be translated to > an "answer", but not every answer is from a 200 OK, so you can't use > that to break other associations. Perhaps this would be a nice addition > to have in the future. I will argue that the only thing that is an answer is a 200. A progress, early media or pre_answer is a 183. Yes both 183s and 200s need to set up media but as you know there could be many 183s and only one 200. If the UA sends an sdp with both the 183 and the 200 would this work correctly? --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
On Sep 29, 2014, at 1:14 PM, Richard Fuchs 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 ends up receiving the call (unless they do a final re-invite). But it should see the 200. Shouldn't that be enough? --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
On Sep 25, 2014, at 10:41 AM, Marino Mileti 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 parallel using alias.. > > - B & C receive the INVITEs & replies with two separate 183 with SDP (in SDP > they specified where they are able to receive audio&video) > > - A receives two 183...& starts to send its RTP video stream to B & C (early > media) > The UA sending the invite can handle two 183s? This is very custom/non-standard. Does this UA also support TURN? Maybe you could do TURN instead. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] R: Re: RTPPROXY & BRANCH
On Sep 25, 2014, at 10:09 AM, Marino Mileti 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 video media > before they answer) > I don't understand. Are you saying that you have clients that when they receive an invite sent video with 183? How do you want to composite the video to show to the caller? It is not RFC3261 compliant to change IP and port from 183 to 200. Of course you can reinvite after the 200. Most B2BUAs require you to ignore early media and generate something locally to send to the caller or just send them 180. Maybe if you explain your use case someone can help you. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio - Separated Servers
On Sep 17, 2014, at 8:13 AM, Bruno Emer wrote: > Hello! > > I am relatively new to Kamailio and I'm trying to create a new enviroment > using it in my company. I am thinking about use Amazon to host the servers > and use OpsWorks to automatically escalate then if necessary. To accomplish > this, my idea is to separate the servers, using one dedicated server to run > as WebSocket, one to run as proxy and one as a Registrar. I'll be using just > one database to store informations to all of my servers. With this, if I need > more resources later, I can just create new servers with the specific roles > (WebSockets, Proxy, Registrar). > > By now, the idea is clear, but the point is that I don't know how to separate > the WebSockets server from the proxy server. Actually, I can do this, but > when I have one agent using a regular softphone and one agent using > WebSockets (with JSSIP) they are not able to establish a session if the > softphone user starts it. Now, I want to know if is there a way to use two > websockets servers, register users using both of then and start sessions > between then, with a separated proxy and registrar. > > Has anyone done this before? Is possible to use kamailio like this? > Most things are possible. There are a few questions you'll want to ask. How reliable does this need to be? How many endpoints and of which type, sip or webrtc? You mention that you have something half working, calls can go one direction. You didn't talk about rtp here. How do you want to handle relaying rtp or will that not be needed and why do you think so? Kamailio is very reliable and can scale vertically before needing to scale horizontally. If you have a small number of clients say less than thousands, I wouldn't bother separating the different functions except for what you need to give you redundancy which isn't happening in the system you describe. Collocating all functions on two servers is a good way to start. You would use srv records for both sip and webrtc endpoints to load balance and failover. The complexity is quite high for a setup like this so one server may offer you enough availability for a time and that time might even be counted in months or years. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 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 rtpproxy -s >> udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2 >> >> Here, rtpproxy bridges between 192.168.2.18 and 127.0.0.1 . >> >> Now I want to migrate to rtpengine with the rtpproxy-ng module in >> kamailio. However, I do not find an equivalent to bridge mode in the >> rtpengine command-line parameters. I see the --ip=IP parameter, but the >> source code expects a single IP address, and cannot be specified more >> than once. The closest I see is the --advertised-ip=IP parameter, but I >> am not sure that it will do what I need. > > Just a quick note to you and anyone else who has asked for this: With > the latest version of rtpengine (git master), bridging between multiple > local interfaces is fully supported. Thank you Richard! --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Minimal configuration without database support
On Aug 27, 2014, at 5:48 AM, Eugene Prokopiev 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 (with its sbc module), in this case. > > I tried to use SEMS but I've got error, which was reported about 2 > years ago without any decision - > http://lists.iptel.org/pipermail/sems/2014-August/004451.html > You will have no problem doing this on Freeswitch. Stripping down FS from it's default config is not hard. I like to rewrite the entire config organizing it for the intended purpose. Thinking through what you want to do, write it down and then rewrite the config. Just because FS has lots of features doesn't mean you need to use them. It's not hard to disable them. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Minimal configuration without database support
On Aug 26, 2014, at 9:46 AM, Eugene Prokopiev 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 Kamailio private address. > You are describing a b2bua, back to back user agent. Freeswitch is what I use. There is SEMS and of course asterisk. I don't recommend asterisk. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine symmetric RTP behavior
On Aug 11, 2014, at 5:04 PM, "Narsay, Deep" 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 recv=40040). > > I am wondering if there is any setting (or flag at compilation time) that > will make it symmetric. > > This rtpengine is daemon only, (not using kernel module). Why do you want rtpengine in front of Freeswitch? Is FS going to give up the media at some point? You should use as few systems passing media as you can. If you don't have disable_rtp_auto_adjust=true on your sofia profile you should be able to receive media there directly from your clients. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Some of the calls drops after 15 minutes + some seconds
On Jul 17, 2014, at 11:27 AM, Andras FOGARASI 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 a TURN server or is it acting more like a media relay where it >> is signaled into the path? What TURN server are you using? How is it >> configured? >> > > The problem occurs even without TURN, in pure peer-to-peer mode. We use > TURN only in emergency case (symmetric NAT and like that...). I do > nothing with session timers - i didn't think about it until now… How do you set up the TURN only in emergency case? Do the phones do it themselves? Does Kamailio control the TURN, rtpengine/mediaproxy-ng? Who sends the bye? --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Some of the calls drops after 15 minutes + some seconds
On Jul 16, 2014, at 4:05 PM, Andras FOGARASI wrote:
> On 7/16/14, 10:00 PM, Frank Carmickle wrote:
>>
>> On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla
>> 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
>>> answered. You can add an xlog(...) at the top of request_route{} and
>>> reply_route{} blocks printing at least the method, call-id, cseq, from and
>>> to header, plus the response code for reply block. In this case you can see
>>> if there is some signaling before call is dropped.
>>
>> Is this happening just on calls between two phones in your domain, or is
>> there a carrier/federation involved?
>>
>
> No other parties are involved, only the two phones involved (and the
> proxy of course).
>
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 a TURN server or is it acting more like a media relay where it is
signaled into the path? What TURN server are you using? How is it configured?
--FC
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Some of the calls drops after 15 minutes + some seconds
On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla 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
> answered. You can add an xlog(...) at the top of request_route{} and
> reply_route{} blocks printing at least the method, call-id, cseq, from and to
> header, plus the response code for reply block. In this case you can see if
> there is some signaling before call is dropped.
Is this happening just on calls between two phones in your domain, or is there
a carrier/federation involved?
--FC
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] TLS and SIP
On May 23, 2014, at 12:43 PM, James Cloos wrote: >>>>>> "FC" == Frank Carmickle 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 packets. > > FC> This is true if you are not using an ephemeral Diffie Hellman cypher > suite. > > Good point. A quick test shows that contacting asterisk-11 over tls/tcp > negotiates rsa key exchange; kamailio does better and agrees to ECDHE-RSA. > > If the trace is of kama talking to asterisk ephemeral is not likely. > Asterisk-12 may be better; I cannot test right now. Nor can I test > freeswitch. > Freeswitch does support most new features of openssl 1.0.1 branch. I believe it defaults to tls1.1 currently but I believe the goal is to only enable tls1.2, with ECDHE+AES128 by default. You can certainly ask it to do what ever openssl supports, except that right now ECDHE is hardcoded to p256. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Modifying SDP in Kamailio
On May 18, 2014, at 2:34 AM, MrIhaveAnOpinionOnEverything wrote: > Hi guys: > > I am a R&D 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 > > > XLITE/LINPHONE ---> KAMAILIO > FREESWITCH > >I am using Freeswitch as a media server. After configuring RTP Proxy and > kamailio to use bridged mode. I was able to successfully setup a voip backend > like the one above. > >I encountered a problem when the UAC I am using is a webclient like sipml5. > >I noticed that when SIP INVITES from KAMAILIO to FREESWITCH are being > passed when a INVITE transaction is initiated from a sipml5 client FREESWITCH > is trying to use the public ip of webrtc server of the sipml5 backend. > Unfortunately, I am using private ip/LAN IP between kamailio and freeswitch. > As a result calls are established but there is no audio that is happening. > I think you're confused, unless I'm confused. What I see from reading the traces is that freeswitch is offering media on a rfc1918 address. You either need to static NAT a non rfc1918 address to freeswitch or allow it to bind one directly. You can use the ext-rtp-ip sofia parameter on your profile if you aren't binding directly. HTH --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] TLS and SIP
On May 22, 2014, at 6:46 PM, James Cloos 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 details. This is true if you are not using an ephemeral Diffie Hellman cypher suite. HTH --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Forward copy of Register information to FreeSWITCH
On Mar 31, 2014, at 7:22 AM, Daniel-Constantin Mierla wrote: > > On 31/03/14 13:07, Alexandr Usov wrote: >> >> >> 2014-03-28 18:16 GMT+02:00 Frank Carmickle : >> >> Freeswitch does not require registration. What are you trying to use >> freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl >> configuration or don't use the directory at all. >> >> >> I want use Kamailio only for registrations and use rtpproxy for registered >> peers. >> So all PBX features I want to use on FS (and/or Asterisk). >> >> So acl style can't serve Voicemail and presence features for not registered >> (on FS) users? > For presence you don't need registration. For example, for MWI just forward > the subscribe request to freeswitch, being sure it matches the > voicebox/voicemail id. > > The default config for kamailio has logic for forwarding mwi subscriptions to > voicemail server. You would need to create the voicemail boxes in the voicemail database for each user. You should do this when creating the user. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Forward copy of Register information to FreeSWITCH
On Mar 28, 2014, at 11:36 AM, Alexandr Usov 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. > > I can't (don't know - how to) use in same way integration with FreeSWITCH. > Can't create in FS directory structure a user with "host=kamailio_ip", FS > require registration. > Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl configuration or don't use the directory at all. HTH --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Freeswitch
On Mar 22, 2014, at 3:18 AM, MrIhaveAnOpinionOnEverything 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 instructions in > http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc > > We are using Kamailio on kamailio 4.0.3 > > and Freeswitch FreeSWITCH Version 1.5.6b > > Everything is working fine with the default setting. But when we > configured TLS what we noticed some errors. > >UAC configured with no TLS calling a UAC configured with TLS works ok. > >UAC configured with TLS calling a UAC configured with TLS gets > disconnected 15-30 seconds after answering the call. > >UAC configured with TLS calling a UAC configured with no TLS gets > disconnected 15-30 seconds after answering the call. > >I cant seem to find any error in the logs or in the ngrep. > Just a stab in the dark here, we really need traces and full configs. Freeswitch sends updates which may not be getting to the user agents when using tls. I'm guessing that freeswitch is trying to send an update directly to the user agents. This could happen if you aren't using recordroute. Turn on siptrace on the FS profile sofia profile $MYPROFILE siptrace on You should see the update go out to the wrong place. It maybe going out to a udp/tcp, not tls, port on the user agent, or it maybe trying to setup a tls socket to a UA but with out a NAT helper to find out the real ip/port. http://kamailio.org/docs/modules/4.0.x/modules/rr.html >Can anyone assist me regarding this issue. Good day. > Hopefully I've pointed you in the right direction. If not maybe I've at least helped you think of something you hadn't thought of yet. Also I recommend that you upgrade to Kamailio and Freeswitch latest. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] New official Debian and Ubuntu repository
On Mar 3, 2014, at 8:16 AM, Victor Seva 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 manage this new system. > > deb.kamailio.org is based on jenkins-debian-glue[2] project running on > AWS EC2 environment thanks to Michael Prokop [3] and myself. All the > needed files, scripts and info to reproduce this system is kept public > at [4]. This is fantastic. Thank you very much for doing this. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RTPProxy/Mediaproxy issue
On Feb 21, 2014, at 4:24 AM, Daniel Grotti 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 in your platform first, and > then in your network devices. > For example, when are you using media-relay, do you receive packets > already jittered ? If yes problem could come from some network devices > in front of you mediaproxy. Are you trying to run on virtualization? You might be having CPU contention issues. --FC ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
