[OpenSIPS-Users] rtpproxy n flag throws "Failed to get dialog"

Hello again :) Rtpproxy documentation says: "for all the calls that require notification, the rtpproxy_engage(), rtpproxy_offer() and rtpproxy_answer() functions must be called with the “n” flag" I'm running rtpproxy_offer("froc") & rtpproxy_response("froc") and opensips 3.1.13, but as soon as I

Re: [OpenSIPS-Users] RTPPROXY / OPENSIPS

Hi Wadii, I haven;t checked the implementation, but the rtpproxy_engage should take care of the 183 with SDP. Have you tested it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online

Re: [OpenSIPS-Users] RTPPROXY / OPENSIPS

I have a lot of calls that should work with RTPPROXY and instead I get dead air. Maybe this is the issue. On Fri, Dec 23, 2022 at 2:45 PM Wadii ELMAJDI | Evenmedia wrote: > > hello , i do have a question related to rtpproxy module documentation. > > The doc describes that rewriting sdp body

[OpenSIPS-Users] RTPPROXY / OPENSIPS

hello , i do have a question related to rtpproxy module documentation. The doc describes that rewriting sdp body should happen during either INVITE , 200 OK or ACK. In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer during the invite and rtpproxy_answer during the 200 OK.

Re: [OpenSIPS-Users] RTPProxy Docker Image

Liviu has done some exploration on getting things handled on Kubernetes. His great presentation is available here: https://youtu.be/JwO0UmauuT4?t=13034 -Max On Wed, Dec 21, 2022, 5:04 PM Terrance Devor wrote: > Hello David, Similar to what we have with LXC > > OpenSIPS - Proxy, Edge Switch,

Re: [OpenSIPS-Users] RTPProxy Docker Image

Hello David, Similar to what we have with LXC OpenSIPS - Proxy, Edge Switch, Managing DIDs and Termination routes, CDR, LB to Asterisk Asterisk - PBX, IVR RTPProxy - Media Relay Everything containerized using docker and deployed to our k8s cluster. I would appreciate speaking to anyone that has

Re: [OpenSIPS-Users] RTPProxy Docker Image

Can you explain more? I.e: params and such? Thanks! On Tue, 20 Dec 2022 at 22:29, Saint Michael wrote: > Opensips+ RTPProxy only works fine with plain LXC containers, > privileged, which basically have access to all the resources of the > box. > That is the model I use with great success. > >

Re: [OpenSIPS-Users] RTPProxy Docker Image

Hello Bret, another approach we are thinking about is put RTPProxy on a VPC. As for Opensips and Asterisk, they can live on a k8s with the understanding that they will not deal with media directly. If anyone can share their experience, I would be interested in hearing from you. Do the RTP guys

Re: [OpenSIPS-Users] RTPProxy Docker Image

Opensips+ RTPProxy only works fine with plain LXC containers, privileged, which basically have access to all the resources of the box. That is the model I use with great success. On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff wrote: > > Hello Terrance, > I wouldn't really recommend this.

Re: [OpenSIPS-Users] RTPProxy Docker Image

Hello Terrance, I wouldn't really recommend this. RTPProxy is going to use a lot of ports in a very large range. That just doesn't work great in docker, but even worse in K8S. I personally would put the RTPProxy outside of K8S. While you might be able to get it to work, you are likely going

Re: [OpenSIPS-Users] RTPProxy Docker Image

Was it something I said? Terrance On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor wrote: > Hello Everyone, > > Wow! Blast from the past... I am a long time member of this list, been a > while. > > Question, anyone successful in deploying RTPProxy to a dockerized > environment? Preferably to a

[OpenSIPS-Users] RTPProxy Docker Image

Hello Everyone, Wow! Blast from the past... I am a long time member of this list, been a while. Question, anyone successful in deploying RTPProxy to a dockerized environment? Preferably to a Kubernetes managed environment. Please Help Team :) Kind Regards, Terrance

Re: [OpenSIPS-Users] Rtpproxy with mhomed option

Hello John, I have missed your answer as it was on the Spam folder. I got it working with rtpproxy_engage(""), but now I can see that the trick was passing the parameter with double quotes. Thank you for your valuable help! Mario On Mon, Feb 22, 2021 at 7:03 AM John Quick wrote: > Hi

Re: [OpenSIPS-Users] Rtpproxy with mhomed option

Hi Mario, I think you are talking about bridged mode for rtpproxy. If so, you will not need to start the rtpproxy daemon with -A, because that is where you have just one interface and you want to masquerade it as a different address. For example if your server is behind NAT, then -l is the actual

[OpenSIPS-Users] Rtpproxy with mhomed option

Hi Everyone, I have been testing opensips with rtpproxy and it works really well on a single interface server and its correct nating. But now that i want to run a configuration with a wan and lan interface the ips at the SDP are not correct. I have seen in some blogs, that they use the flags (ie)

Re: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

Provider -- -Original Message- From: Eugene Christensen Sent: Thursday, February 11, 2021 11:21 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How

Re: [OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

at +1 (801) 287-9419, and destroy the original transmission and its attachments without reading them or saving them to disk. -Original Message- From: Users On Behalf Of Rick McGill - ? Sent: Thursday, February 11, 2021 8:56 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users

[OpenSIPS-Users] RTPproxy for OpenSIPS 3.1 on Debian 10.7 (How To)

Dear OpenSIPS Community, Yeah I'm a newby to OpenSIPs app specifically and I just realized that I need a separate install for RTPproxy for my OpenSIPS. My question is there a repo for that or do I need to download and install it manually? Also I cannot find and good material for how to install

Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

Hi Tomi, The "opensipsctl fifo rtpproxy_show" command does not give a 'status', but a 'disabled' value - for example: Set:: 0 node:: udp::7898 index=0 disabled=0 weight=1 recheck_ticks=0 node:: udp: :7898 index=1 disabled=0 weight=1 recheck_ticks=0 node:: udp: :7898

Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

Hi, I believe MU is just typo should be MI The ’status’ should also be retrieved differently not with the $ opensipsctl fifo rtpproxy_show command I think you can get the ’status’ directly from the database with SQL query. Tomi On 24. Jun 2020, at 16.33, solarmon wrote:  Hi

Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

Hi Bogdan-Andrei, There is only 'Memory State' in CP. There is no 'status' in CP. Sorry, what is 'MU'? I'm using opensips 2.4.x if that makes any difference. Thank you. On Wed, 24 Jun 2020 at 13:49, Bogdan-Andrei Iancu wrote: > Hi, > > In CP, the `status` is the status from the SQL table

Re: [OpenSIPS-Users] rtpproxy 'Memory State' 'status'

Hi, In CP, the `status` is the status from the SQL table and the `memory status` is the status provided by the MU rtpproxy_show. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 6/24/20 12:16 PM, solarmon wrote: Hi, The command

[OpenSIPS-Users] rtpproxy 'Memory State' 'status'

Hi, The command "opensipsctl fifo rtpproxy_show" does not return the 'status' of the rtpproxy node. In OpenSIPS Control Panel, the RTPProxy table has a '*Memory State*' column which seems to be the 'status' of that node. How can I get this 'Memory State' 'status' in command line form so that it

Re: [OpenSIPS-Users] [RTPproxy] catch_dtmf module has landed in rtpproxy/master

Thanks, Gohar! Very good questions: 1. Module relies on RFC2833 (DTMF Events). No in-band decoding is implemented at the moment. 2. Module is observer-only for now. It doesn't try to block or in any way alter the RTP stream being forwarded. Hovewer, that might be something to consider as a next

Re: [OpenSIPS-Users] rtpproxy module not supporting valid payload types

Yes, for sure. As long as the transport is UDP based, the RTPProxy would just work. The change should be trivial, you can get it fixed locally, test and then open a pull request against opensips repo. -Max On Thu., Apr. 2, 2020, 11:43 a.m. Robert Dyck, wrote: > Regarding opensips-3.0 > > Use

[OpenSIPS-Users] rtpproxy module not supporting valid payload types

Regarding opensips-3.0 Use case - webrtc client behind NAT The rtpproxy module emitted the error message "can't extract media port from the message" ( by the way, very misleading ). In reality extract_mediainfo fails because it could not find a supported payload type in the media description.

Re: [OpenSIPS-Users] rtpproxy status health checking

Hi Solarmon, I can't comment on the exact behaviour internally regarding the ticks value however I thought I could share it as I see it from a user perspective. The relevant settings I use are as follows: modparam("rtpproxy", "rtpproxy_disable_tout", 60) modparam("rtpproxy", "rtpproxy_timeout",

Re: [OpenSIPS-Users] rtpproxy status health checking

Hi, Just an update/correction. I notice that when I make a call through opensips, the recheck_ticks value does reduce slightly to 928855, but it seems to stay at that for subsequent calls. In Control Panel. The memory state does turn to Green. # opensipsctl fifo rtpproxy_show Set:: 0

Re: [OpenSIPS-Users] rtpproxy status health checking

Hi Razvan, I'm using opensips 2.4.6 (x86_64/linux) so I don't think opensips-cli is available? I'm using opensipsctl to show the rtpproxy status. This is the output of the command after I have turned off rtpproxy with index 0: # opensipsctl fifo rtpproxy_show Set:: 0 node:: udp:

Re: [OpenSIPS-Users] rtpproxy status health checking

Hi, Solarmon! The parameter you should use is exactly the one you are using, rtpproxy_disable_tout[1]. That parameter says that after OpenSIPS detects the node as being down, it re-tries to send them requests after 20 seconds (according to your configuration). Are you checking the rtpproxy

[OpenSIPS-Users] rtpproxy status health checking

Hi, Can somebody clarify when the rtpproxy status and health checks are done and what configuration is required. I am finding that the status/health of an rtprpoxy node is only done/checked during opensips startup or rtpproxy module config reload. If the rtprpoxy node goes down or comes back up,

Re: [OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

Mark, You can detect if the INVITE came from your Asterisk by testing the $si pseudo-variable. That will allow you to identify the direction of the call. I usually set a flag for this purpose. For example: If ($si == "my.ast.er.isk") setflag(DIR_OUT); At the point where you engage the

[OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

Hello everyone, all help gratefully received, I've been slogging away at this for ages! I have OpenSIPS 2.4.4 & RTPProxy behind 1:1 NAT's (different hosts). RTPProxy runs so: /usr/local/bin/rtpproxy -s unix:/var/run/rtpproxy/rtpproxy.sock -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s

Re: [OpenSIPS-Users] RTPPROXY ENGAGEMENT

If you are experiencing double sdp re-wites it means that you are engaging rtpproxy more then once. Add some xlogs in your script where you engage rtpproxy and figure out why it is engaged twice. Regards, Ovidiu Sas On Fri, Jan 25, 2019 at 12:57 PM Mark Thomas wrote: > > I have an issue and I

[OpenSIPS-Users] RTPPROXY ENGAGEMENT

I have an issue and I don’t know how to resolve it. I’ve got a 486 route that has to engage rtpproxy to bridge to another network. My problem is I have to engage from UAC to UAS on the public side before it sends to voicemail. Whenever I attempt to engage rtpproxy on the leg going to the

[OpenSIPS-Users] RTPPROXY not starting on correct user account

Greetings, I am having trouble with RTPProxy bridging media in Opensips 2.3.5 Currently, it seems that RTPProxy is starting as user "root" and not user "rtpproxy" as it should. In etc/init.d/rtpproxy I have added the additional daemon options to load the service as user "rtpproxy", but these

Re: [OpenSIPS-Users] rtpproxy error messages

Hi, John! Those errors are reported by the RTPProxy server, and depending on the error, they can be handled by opensips. In your case, the errors 71 and 72 are reported by RTPProxy when it can not create UDP listeners for media. Usually these errors are triggered when rtpproxy runs out of

[OpenSIPS-Users] rtpproxy error messages

Hello, Can anyone help me understand what might be causing the following error messages to be reported by OpenSIPS? 2018-01-16 16:42:47 ERROR:rtpproxy:engage_rtp_proxy4_f: error forcing rtp proxy 2018-01-16 16:42:48 ERROR:rtpproxy:force_rtp_proxy_body: unhandled rtpproxy error: 71 2018-01-16

Re: [OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

I don't think restarting RTPProxy is acceptable, because you will loose the ongoing RTP sessions. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 01/15/2018 01:03 PM, Adrian Fretwell wrote: Razvan, Thankyou for clarifying this, I can work around it either with

Re: [OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

Razvan, Thankyou for clarifying this, I can work around it either with VIP or by restarting RTP Proxy with a different -n value. Kind regards, Adrian. On 15/01/18 09:02, Răzvan Crainea wrote: Hi, Adrian! Unfortunately RTPProxy can send timeout notifications only to one timeout_socket -

Re: [OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

Hi, Adrian! Unfortunately RTPProxy can send timeout notifications only to one timeout_socket - the one specified in the -n parameter. We did discuss with Maxim a while back the ability to support different timeout_sockets to different opensips instances, but that discussion didn't materialize

[OpenSIPS-Users] RTPProxy Timeout socket more than one SIP proxy

Hello & Happy New Year, Just trying to work out what to do with the timeout notifications from RTPProxy when the RTPProxy is used by more than one Opensips proxy. RTPProxy manual says: /-n timeout_socket / /This parameter configures the optional timeout notification socket. The

Re: [OpenSIPS-Users] rtpproxy socket

Hello, yes sure. 1. On initial INVITE store rtpproxy socket to any dialog value (store_dlg_value)2. In dialplan store rtpproxy socket (as match_exp) with attrs field, in which sock_id of rtpproxy has been inserted.3. Insert additional records to rtpproxy_sockets table with individual sock_id

Re: [OpenSIPS-Users] rtpproxy socket

HiWould you mind to share your experience with us?Thanx.-- Wbr, Serge via mobile12.09.2017, 10:20, "Denis via Users" :Hello! I found solution myself. The question is closed. -- С уважением, Денис.Best regards, Denis 08.09.2017, 14:51, "Denis via Users"

Re: [OpenSIPS-Users] rtpproxy socket

Hello! I found solution myself. The question is closed. -- С уважением, Денис.Best regards, Denis 08.09.2017, 14:51, "Denis via Users" :Hello! Opensips 2.3 One interesting question about rtpproxy using. I can use multiple rtpproxy through one set. During each call

[OpenSIPS-Users] rtpproxy socket

Hello! Opensips 2.3 One interesting question about rtpproxy using. I can use multiple rtpproxy through one set. During each call Opensips chooses certain socket inside the certain set of proxies.Till this time everything clear.But during existing session SIP UA (caller or callee never mind) can

Re: [OpenSIPS-Users] rtpproxy not relaying data

a lot! Sent with [ProtonMail](https://protonmail.com) Secure Email. Original Message Subject: Re: [OpenSIPS-Users] rtpproxy not relaying data Local Time: 22 de febrero de 2017 6:14 AM UTC Time: 22 de febrero de 2017 10:14 From: raz...@opensips.org To: users@lists.opensips.org

Re: [OpenSIPS-Users] rtpproxy not relaying data

Hi, Waldo! I only see the command for Update (initial request), I don't see the command for Lookup(200 OK). Moreover, are you sure RTP traffic gets to the rtpproxy machine? Because RTPProxy statistics doesn't see any packets getting to the server. Can you double check if the firewall allows

[OpenSIPS-Users] rtpproxy not relaying data

Hi: I am trying to use rtpproxy with my SIP proxy. It starts a session and both SIP clients send data to rtpproxy port as I see on wireshark after I modify the SDP part. I see data comming to rtpproxy but no data comes out of it. What could be going on wrong? Configuratoin issue? I send here

Re: [OpenSIPS-Users] rtpproxy and record calls

Hello Razvan! I am sorry for long answer, only now could return to these questions. So 1) I run rtpproxy_start_recording() for both INVITE and 200 OK.The some content of the opensips.cfg"...modparam("rtpproxy", "rtpproxy_sock", "1 == udp::221") route[1]if (is_method("INVITE")&&!has_totag()) {  

Re: [OpenSIPS-Users] rtpproxy and SRTP

ons I hope there will be no problem. I am setting up a test environment at the moment. John Quick Smartvox Limited From: Sasmita Panda [mailto:spa...@3clogic.com] Sent: 07 February 2017 06:54 To: john.qu...@smartvox.co.uk; OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS

Re: [OpenSIPS-Users] rtpproxy and SRTP

Hi , I have tested 1 scenario . If there is two end point which support encrypted media (SRTP) and there is rtpproxy between them . Then rtpproxy works as usual . It update the C line Ip in the SDP and forwards the request and response . This is what I tested . And its working .

[OpenSIPS-Users] rtpproxy and SRTP

Please, does anyone know if rtpproxy works with SRTP? John Quick Smartvox Limited ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Rtpproxy bug

Hi Robert, It looks like a bug to me, but you have to check and report this to someone having commit rights on the rtpproxy repository. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02/04/2017 01:59 AM, Robert Dyck wrote: I am

Re: [OpenSIPS-Users] rtpproxy and record calls

Hi, Denis! Regarding 1, did you try to run rtpproxy_start_recording() for both INVITE and 200 OK? Regarding 2, '/' is a valid character in Call-id. Therefore the problem is at the RTPProxy side - before writing the CDR they should escape (or transform somehow) the '/' character in something

Re: [OpenSIPS-Users] rtpproxy and record calls

Hello!Is there any information about the problem?Thank you.-- С уважением,Путято ДенисBest regards, Denis14:47, 27 декабря 2016 г., Denis via Users :Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from

[OpenSIPS-Users] rtpproxy and record calls

Hello! I try to use rtpproxy for call recording and have two problems 1) rtpproxy records only one way of the call (from callee to caller). For starting rtp proxy i use rtpproxy_engage("conrf",,"1",) function. 2) i am using top_hiding with "C" flags (which should change callid) and i noticed, that

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Hello, Flavio! Thank you very much for your help! I made some test and it worked.But, in additional, i want to ask you about g729 codec. In rtpproxy documentation says that beginning from 2.0 g729 codec supported.In the dictionary of extractaudo i see some g729* files. Is there necessary to

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Hi, Yes you can extract audio from rtpproxy. The extractaudio utility is very handy and you can compile with G.729 from the linphone project bcg729. It is very easy to use, simply use the utility followed by the name of the recording without any extension. Check the source code for the other

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

I read about it, but if i need extract g729?-- С уважением,Путято ДенисBest regards, Denis18:08, 23 декабря 2016 г., Ovidiu Sas :Here are the steps to extract the audio:https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-recorded-with-rtpproxy/Regards, 

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Here are the steps to extract the audio: https://voipembedded.wordpress.com/2011/11/15/extracting-audio-from-calls-recorded-with-rtpproxy/ Regards, Ovidiu Sas On Dec 23, 2016 09:52, "Denis via Users" wrote: Hello, Bogdan! I mean haw can i extract audio from .rtp

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Denis, as I see, this is a tool provided by the rtpproxy project - you should dig and ask more on their side. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 23.12.2016 16:51, Denis wrote: Hello, Bogdan! I mean haw can i extract audio from

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Hello, Bogdan! I mean haw can i extract audio from .rtp file, which was recorded by rtpproxy.In documentation to rtpproxy said, that extractaudio utility can do that. But how do that, i cannot find. -- С уважением, Денис.Best regards, Denis   23.12.2016, 17:49, "Bogdan-Andrei Iancu"

Re: [OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Hi Denis, You mean to record the RTP going through rtpproxy ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 23.12.2016 12:22, Denis wrote: Hello! I want to ask about extractaudio. Is there any manual for it? How can i use it? Has anybody

[OpenSIPS-Users] rtpproxy 2.0 and extractaudio

Hello! I want to ask about extractaudio. Is there any manual for it? How can i use it?Has anybody operation experience of this soft? Thank you. -- С уважением, Денис.Best regards, Denis    ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] rtpproxy and timeout socket

Yes, It seems OpenSIPS is not compatible with rtpproxy 2 version for timeout notifications. That's because OpenSIPS always strips the tcp: from the beginning of the socket, while RTPProxy 2 always waits for this prefix. Please open a bug/feature request on github for us to track this issue:

Re: [OpenSIPS-Users] rtpproxy and timeout socket

Hello, Razvan! 2.2.alpha.20160822 -- С уважением, Денис.Best regards, Denis 21.12.2016, 11:58, "Răzvan Crainea" :Hi, Denis!What version of rtpproxy are you using? There might be an incompatibility issue here.Best regards,Răzvan Crainea OpenSIPS Solutions

Re: [OpenSIPS-Users] rtpproxy and timeout socket

Hello, Razvan! 2.2.alpha.20160822 -- С уважением, Денис.Best regards, Denis 21.12.2016, 11:58, "Răzvan Crainea" :Hi, Denis!What version of rtpproxy are you using? There might be an incompatibility issue here.Best regards,Răzvan Crainea OpenSIPS Solutions

Re: [OpenSIPS-Users] rtpproxy and timeout socket

Hi, Denis! What version of rtpproxy are you using? There might be an incompatibility issue here. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 12/21/2016 07:37 AM, Denis wrote: Hello! I am using rtpproxy in my VoIP network and now i want to locate it on

[OpenSIPS-Users] rtpproxy and timeout socket

Hello! I am using rtpproxy in my VoIP network and now i want to locate it on another server rather then Opensips instance.Rtpproxy is started with such parameters:/usr/local/rtpproxy2/bin/rtpproxy -u -p -l -s udp:: -i -T 10 -n tcp::2229 -m 35000 -M 45000 -r -R -P -d INFO LOG_LOCAL5 The Proxy

Re: [OpenSIPS-Users] rtpproxy compile

Just one additional question.Is there any documentation about extractaudio utility?I cannot find any README file about it in the package and http://www.rtpproxy.org/post/v2release/ just mention about it. -- С уважением, Денис.Best regards, Denis 20.12.2016, 11:15, "Răzvan Crainea"

Re: [OpenSIPS-Users] rtpproxy compile

Hello, Razvan! Thank you very much for your help! You are right, i should make additional ./configure after libsndfile installed. Now everything work fine! Thank you.  -- С уважением, Денис.Best regards, Denis 20.12.2016, 11:15, "Răzvan Crainea" :Hi, Denis!Can you run

Re: [OpenSIPS-Users] rtpproxy compile

Hi, Denis! Can you run 'ldconfig -p | grep sndfile'? Do you see the .so libraries in the output? If not, perhaps you should run 'ldconfig'. If this still does not work, you should manually add the library's directory in the library path[1]. If you do get sndfile in the output, but it still

[OpenSIPS-Users] rtpproxy compile

Hello! I want to use extractaudio utility to extract audio and write it to some .wav file.But, during compiling of this utility i get such error"extractaudio.c:325: undefined reference to `sf_open'extractaudio.c:389: undefined reference to `sf_write_short'extractaudio.c:400: undefined reference to

Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

Hi, Robert, Does rtpproxy _autobridge work? http://www.opensips.org/html/docs/modules/1.11.x/rtpproxy#id293590 Regards, xiaofeng ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

Hi, Robert! Yes, in cases where you don't need IPv6, use II for those requests. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/09/2016 07:12 PM, Robert Dyck wrote: I should have described the scenario in more detail. The rtproxy is in bridge mode because

Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

Hi, Robert! Yes, the I and E parameters are mandatory, and they should describe how the RTP will flow. For example if the flow is from IPv4 to IPv6, you should use EI; if the flow is from IPv4 to IPv6, then you should use IE. And so on, depending on the call flow. Regarding the address

Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

Thank you Assuming rtpproxy was started with IPV4 as the first address and IPV6 as the second, then in the NAT scenario, are the II flags mandatory in offer/answer? Slightly off topic, what sort of scenario would require the address parameter for offer/answer? On November 8, 2016 09:57:30 AM

Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

Hi, Robert! See my answers inline. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 11/08/2016 02:15 AM, Robert Dyck wrote: I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6 interworking. The articles I have read say that you need to

[OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6 interworking. The articles I have read say that you need to assign an address from each address family to rtpproxy. They go on to say that rtpproxy will then be in bridged mode. Others define bridge mode as assigning

Re: [OpenSIPS-Users] rtpproxy TCP connection

Could this also explain why other TURN servers don't work so well over TCP with OpenSIPS? On 18 February 2016 at 12:09, Răzvan Crainea wrote: > Hi Gomtesh! > > Currently only UDP and UNIX datagrams sockets are supported. > > Best regards, > Răzvan > > > On 02/18/2016 01:15

Re: [OpenSIPS-Users] rtpproxy TCP connection

Hi Gomtesh! Currently only UDP and UNIX datagrams sockets are supported. Best regards, Răzvan On 02/18/2016 01:15 PM, Gomtesh Jain wrote: Is it possible to make tcp connection to rtpproxy from opensips ? I am trying modparam("rtpproxy", "rtpproxy_sock", "tcp:127.0.0.1:2

[OpenSIPS-Users] rtpproxy TCP connection

Is it possible to make tcp connection to rtpproxy from opensips ? I am trying modparam("rtpproxy", "rtpproxy_sock", "tcp:127.0.0.1:2") But it is not working . any suggestions ? Thanks, Gomtesh ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] rtpproxy and parallel forking

Hi Pete, I assume you do rtpproxy_answer() for the 200 OK on B leg, right ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 23.09.2015 11:44, Pete Kelly wrote: I am using rtpproxy with parallel fork and noticed some interesting behaviour

Re: [OpenSIPS-Users] rtpproxy and parallel forking

Hi Pete, To the best of my knowledge no rtp proxy: mediarelay, rtpengine, rtpproxy deals with forking and early media "well". I believe this is more a failing of the 183 draft than anything else. For example If I parallel fork a call to A and B, A sends 183 with an IVR but then B sends a

[OpenSIPS-Users] rtpproxy and parallel forking

I am using rtpproxy with parallel fork and noticed some interesting behaviour (by rtpproxy). If the INVITE is forked to 2 destinations (A and B), one of them (A) may send a 183 with media, meaning there is media being sent to the rtpproxy. However if it is B that answers, rtpproxy will still

[OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Hi, I would like to know which is more effective for NAT traversal, rtpproxy or STUN/TURN/ICE implementation. I heard that TURN server with one public IP can function equivalent to rtpproxy, and TURN server with two public IPs is more effective than rtpproxy. Is that true?

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Stun/Turn/Ice are usefull where Client is behind a NAT and OpenSIPS has public IP. You could use nathelper modules instead of Stun, to set the right IPs in the messages from client. If OpenSIPS is behind a NAT too (it has private IP) you must use RTPProxy too with a proper configuration. Il

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

That said, only clients that supports turn will use it, check your clients features. Rtpproxy, mediaengine, and the like do not rely on clients support, they are.enforced by sip proxy manipulation of sdp. On 29 Aug 2015 17:02, Giovanni Maruzzelli gmar...@gmail.com wrote: Stun/turn are the only

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Both will work. You can check other aspects inherently to your project and implementation: performances, integration, etc Rttproxy, media engine and the like can give you more services related to the fact they are controlled by the proxy. sent from my mobile, Giovanni Maruzzelli cell: +39 347

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Stun/turn are the only methods used by webrtc peers, and because are used through ICE they're very effective. You can check coturn for an advanced implementation. That said, only clients that supports turn will use it, check your clients features. Rtpproxy, mediaengine, and the like do not rely

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Sorry previous message I sent was meant to be a quote. All my clients will use the same UAC which supports ICE/TURN, so that is not an issue. I just want to know which is more effective solely on the basis of NAT traversal ability. On 29 Aug 2015 18:01, Nabeel nabeelshik...@gmail.com wrote:

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Thanks, but I'm still looking for a more direct comparison of rtpproxy vs. TURN/ICE only based on their effectiveness, nothing else. I know both work but I would like to know of any evidence that TURN with two public IPs is more effective than rtpproxy alone. On 29 Aug 2015 18:36, Giovanni

Re: [OpenSIPS-Users] [RTPproxy] Re: Announcing rtpproxy v2.0.0

Pull requests are welcome, guys. We are hitting 30k rtp sessions on one of our largest clusters. Lot of changes are coming into git repo soon to make rtp_cluster component run smoothly under such conditions. On Mar 16, 2015 8:56 PM, John Mathew john.mat...@divoxmedia.com wrote: Yes On Tuesday,

Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

Hi, Marco! From RTPProxy point of view, you can't differentiate between SIP replies, because for all of them you call the same function - rtpproxy_answer(). Now, if the client decides to send RTP for 183 (and indeed, I've seen this several times), there's not that much that you can do.

Re: [OpenSIPS-Users] Rtpproxy issue with serial forking

Hi, John! I guess you are only using rtpproxy when the caller is behind NAT. In this case, you don;t have to detele at all the rtpproxy session, simply call rtpproxy_answer() on all replies. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/21/2015 01:05 PM,

Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

: Donnerstag, 22. Januar 2015 09:36 An: users@lists.opensips.org Betreff: Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK Hi, Marco! From RTPProxy point of view, you can't differentiate between SIP replies, because for all of them you call the same function - rtpproxy_answer

Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

=NDdhYjlhY2Y2NDM5NzJkNjM4NjgzZDIwZTljYTc4YTQ Von: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] Im Auftrag von symack Gesendet: Mittwoch, 21. Januar 2015 17:11 An: OpenSIPS users mailling list Betreff: Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK Can you

Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

Can you please post where you are using rtpproxy_offer/_answer?​ Nick from Toronto. ​ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Rtpproxy issue with serial forking

I have used opensips+rtpproxy for years for simple scenarios but now I am trying to use it with serial forking. My flow is UA---Invite ---Opensips 1 Branch ---Media Server

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