Re: [OpenSIPS-Users] RTPENGINE flag (substitute)

Hi, Alain! I don't think there is a flag that can be passed to rtpengine to change this, mostly because I don't think there is much RTPEngine can do to "support" the change - if it receives the 16'th DTMF, what should it do, just drop it? But if you want/need a correct answer, you should be a

[OpenSIPS-Users] RTPENGINE flag (substitute)

Hello all, I know I'm not on an rtpengine support group, but the atmosphere is nicer here ;) is that someone could give me the correct syntax of the flag to give to rtpengine to convert: from : a=fmtp:96 0-16 to a=fmtp:96 0-15. The rtpengine documentation is not clear about the subst

Re: [OpenSIPS-Users] rtpengine and multiple instances

Thanks, sounds like it's time to upgrade 😊 BR/OLle -Original Message- From: Users On Behalf Of Razvan Crainea Sent: den 6 oktober 2023 13:13 To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpengine and multiple instances Hi, Olle! Yes, the offer&answer should be

Re: [OpenSIPS-Users] rtpengine and multiple instances

Hi, Olle! Yes, the offer&answer should be taken by one single node. How it internally works is we compute a hash of the callid of the call and based on that hash, and we consider that hash a random value to pick one available server. This means that as long as the available list does not chan

[OpenSIPS-Users] rtpengine and multiple instances

Hi we have seen an issue with rtpengine module in 2.4. We ran a setup with two opensips edge proxies each having a local rtpengine running on it. When we start using the rtpengines in a cluster, we saw that some commands perhaps in 1-2% of the calls are send to the wrong rtpengine:

Re: [OpenSIPS-Users] rtpengine not kernel packet forwarding

Oh wow, it was just a case of hanging on the test call for a bit longer - thank you so much! I never saw that crucial bit of info in any of the guides I found and it's perhaps a bit odd that the distro scripts added the 1st and 3rd (as quoted below) rules to iptables but didn't add the last one.

Re: [OpenSIPS-Users] rtpengine not kernel packet forwarding

The kernel module still requires a fallback user space rule for the first few packets. Once the user space daemon determines the forwarding rule set, it pushes to the kernel module. The kernel module can then pickup the packets. Their GitHub page has a detailed step by step for this process. -

[OpenSIPS-Users] rtpengine not kernel packet forwarding

Hi, I realise this is an OpenSIPS list and not an rtpengine list, but I'm stumped so am wondering whether anyone on here can shed any light on the issue. I have rtpengine working with OpenSIPS but if I try to use kernel packet forwarding it doesn't forward the packets. lsmod shows xt_RTPENGINE a

[OpenSIPS-Users] rtpengine session-name

Hi, I'm using opensips 3.3.1 with rtpengine, but I don't see replace-session-name. Is this flag implemented? Thanks A ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RTPEngine log

Just to compare.   This call seems to be OK. Another order of log entries. «using iLBC/8000» 2 lines. And an «Average MOS» results (no such like in previous log).   10.45.145.33 — asterisk 10.45.144.69, 195.209.XXX.YY — opensips/rtpengie (int, ext) 217.66.158.139 — cell phone provider nat box   11

[OpenSIPS-Users] RTPEngine log

Hi list, I would like to discuss RTPEngine logs, to know your opinion if such log is normal or not.   The client complained that haven’t heard anything during the call.   10.45.145.33 is Asterisk 195.209.XXX.YY is OpenSIPS + RTPEngine 217.66.157.207 is cell phone network from which an UAC connects

Re: [OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

Hey Mark, The load balancing weights are set on a per node basis via their socket URL. If no weight is explicitly set, then the default is 1. schema: ::(=) ex: udp:192.168.1.200:2=25 There currently is no way to dynamically change the weight of a node, although there's an open PR [1] wh

Re: [OpenSIPS-Users] rtpengine weight configuration

Hi Jeff, If I'm not wrong, the syntax is similar to rtpproxy, so try like: "udp:localhost:12221=1 udp:localhost:1=2" to get 1:2 weight ratio. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 2021 https://opensips.org/tr

[OpenSIPS-Users] rtpengine weight configuration

Hello, This is on OpenSIPS 2.4, although I suspect the answer will be the same for any modern-ish OpenSIPS version. I have a collection of rtpengine entries loaded into the rtpengine table. All works well. The "rtpengine_show" fifo command lists them all with weight=1. If I'd like to load some

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

Just posting this to document the end result. The following line to pass the transcoding parameters I wanted directly to the RTPEngine daemon worked for me: transcode-opus/48000/2/32000//stereo--0;useinbandfec--1;maxplaybackrate--16000;sprop-maxcapturerate--16000 NB - The '--' part means '=' and

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

No worries On Wed, Sep 15, 2021, 11:43 Mark Allen wrote: > Ah! Thanks Răzvan and Johan. I was thinking that the options described in > the OpenSIPS RTPEngine module documentation were the only ones > available, but I missed the key phrase... > > *When passing an option that OpenSIPS is not a

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

Ah! Thanks Răzvan and Johan. I was thinking that the options described in the OpenSIPS RTPEngine module documentation were the only ones available, but I missed the key phrase... *When passing an option that OpenSIPS is not aware of, it will be blindly sent to the rtpengine daemon to be proces

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

I think the proper way to do it is `transcode-opus/48000/2/32000//maxplaybackrate--48000;stereo--0;useinbandfec--1;maxaveragebitrate--32000` Anyway, as Johan suggests, follow the rtpengine documentation, you will find more info there. Best regards, Răzvan Crainea OpenSIPS Core Developer http:/

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

Follow the rtpengine documentation. The necessary flags are given there, just pass them in an avp to rtpengine_offer or answer On Tue, Sep 14, 2021, 14:55 Mark Allen wrote: > Hi Răzvan > > OpenSIPS passes the commands to RTPengine as you supply them, so in >> theory this should be supported. Did

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

Hi Răzvan OpenSIPS passes the commands to RTPengine as you supply them, so in > theory this should be supported. Did you try to set this up? Can you > provide any logs about this? > > No. I'm not sure how I would specify this in the opensips.cfg rtpengine_offer parameters. From the documentation f

Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

Hi, Mark! OpenSIPS passes the commands to RTPengine as you supply them, so in theory this should be supported. Did you try to set this up? Can you provide any logs about this? Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 9/14/21 14:01, Mark Allen

[OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

Hi all, I'm using RTPEngine for transcoding between Opus and PCMU. According to the RTPEngine documentation (https://github.com/sipwise/rtpengine), it can accept formatting parameters for CODECs that support them. For instance, Opus allows you to change the maxaveragebitrate. However, in the OpenS

Re: [OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

Thanks John. That's really helpful On Fri, 20 Aug 2021, 14:40 John Burke via Users, wrote: > Hey Mark, > > The load balancing weights are set on a per node basis via their socket > URL. If no weight is explicitly set, then the default is 1. > > schema: > ::(=) > > ex: > udp:192.168.1.200:2=2

Re: [OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

This may be a good place to ask this, or perhaps not. Is it possible to define the DB rows in such a way that causes a priority-based behavior? Prefer this one, move to the next if the first isn't available, etc. As I think through it, it may be possible to do that manually in the script by call

Re: [OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

Hey Mark, The load balancing weights are set on a per node basis via their socket URL. If no weight is explicitly set, then the default is 1. schema: ::(=) ex: udp:192.168.1.200:2=25 There currently is no way to dynamically change the weight of a node, although there's an open PR [1] wh

Re: [OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

I've not been able to find the answer to this. Can anyone help? On Thu, 22 Jul 2021 at 11:02, Mark Allen wrote: > In the rtpengine documentation [1] in the section "1.2 - Multiple RTP > proxy usage" it says... > > "The balancing inside a set is done automatically by the module based > on the

[OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

In the rtpengine documentation [1] in the section "1.2 - Multiple RTP proxy usage" it says... "The balancing inside a set is done automatically by the module based on the weight of each RTP proxy from the set." ...how is the weighting determined? Is there a parameter to allocate a weighting

[OpenSIPS-Users] RTPENGINE RTP-SRTP

Hi. I’m trying to use RTPENGINE + OPENSIPS to connect a RTP PBX with a External SRTP. For this I’m using RTPENGINE module. Questions. Is there a way to indicate to RTPENGINE to use certain cypher suite?. Now, I hace in the offer : m=audio 62768 RTP/SAVP 120 9 0 8 18 101 a=rtpmap:120 opus/480

Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

Hello Alex, I tried the flag loop-protect and it works. I get the following message on the logs: rtpengine[22968]: INFO: [fd72c95927567a5b]: Ignoring message as SDP has already been processed by us Thanks for your inputs! Regards On Fri, Jul 17, 2020 at 4:15 PM Alex Balashov wrote: > I thi

Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

I think one angle on the problem is to identify why RTPEngine is being invoked twice for the same SDP (most likely the same SIP message). That's best done by looking at the content of the reinvite itself. Focus especially on the Request URI and the Route header. I am not sure if the RTPEngine

Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

Thanks for your explanation Alex, Actually i compiled the latest..git clone https://github.com/sipwise/rtpengine.git But still i dont have a clue on how to solve it, any advise? Thank you Mario On Fri, Jul 17, 2020 at 2:34 PM Alex Balashov wrote: > This happens when an SDP body that has alrea

Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

This happens when an SDP body that has already been passed to RTPEngine, and already adulterated by RTPEngine, is passed to it yet again. Newer versions of RTPEngine have a loop protection feature to deal with it. It involves injecting an unregistered a=rtpengine attribute into the SDP, to say

[OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

Hello Everyone, I have a scenario, where opensips is working with rtpengine . Call connects fine with audio. I have enabled reinvite pinging to keep the call alive ; when the reinvite is sent to both legs of the calls i get error messages and NO AUDIO. The reinvite has full SPD body. some con

Re: [OpenSIPS-Users] Rtpengine configuration problem

Answering my own question, it appears that rtpengine reverses the logic of rtpproxy with regard to the media source address. It trusts the "c" ( connection ) address that the UA puts there. In this case the UA had learned what its public address was. Rtpengine trusts this instead using the so

[OpenSIPS-Users] Rtpengine configuration problem

While configuring my script for rtpengine I got a rather confusing result. The test involved a UA tethered to a phone with only IPV4 availble. The test was a call to a UA registered with an IPV6 address. The call was answered successfully but there was no media. A sniffer at the rtpengine hos

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hi, Ovidiu! I agree, the fact that no one reported it before is a bit interesting. On the other hand, not sure if >1000 buffers is expected, perhaps the flags you are using for rtpengine are increasing the number of buffers. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.ope

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hi Razvan, I have updated the rtpproxy module and everything is backported down to 2.4. Detecting IOV_MAX would be kind of an overkill ... The error logs were improved and the error is now searchable in the mailing list. If everyone stumble across it, it should be easy to troubleshoot. It seems t

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hi, Ovidiu! I agree we should match the two modules to use a common logic. I believe we should backport this in 2.4 as well, since it doesn't work for all scenarios, therefore it is considered a bug. I somehow doubt this should be a module's param, because it is only needed in certain circums

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Fix implemented on trunk, 3.1 and 3.0. We should update the fix on rtpproxy to match rtpengine and move the OSIP_IOV_MAX define somewhere upper in the code tree to be visible by both rtpproxy and rtpengine modules. For trunk, maybe we should add a new param. Should we backport this fix to 2.4? Th

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hello Razvan, On ubuntu we have a failure for more then 1024 buffers. There is something defined in some experimental headers (as __IOV_MAX), but the IOV_MAX define is not visible in the rtpengine module. And yes, there is an extra buffer. I guess the length of that extra buffer was zero and that

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hi, Ovidiu! So you're saying that the IOV_MAX is not explicitly defined, but it does fail after 1024 buffers, correct? If so, perhaps we should limit the number of buffers to 1024, if not already defined. I did notice the extra vcnt as well, but I though that was related to the fact that it al

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

It looks like the hidden IOV_MAX is set to 1024 in debian. I tested a patch and all looks good. It seems that there is an error in the code: writev has the wrong number of iovcnt (should be one less). I tested and all looks ok (the code in rtpproxy.c has the proper iovcnt). -ovidiu On Thu, Jun 25

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

We will need to add a param to control the max number of buffers. -ovidiu On Thu, Jun 25, 2020 at 1:02 PM Ovidiu Sas wrote: > > It seems that when we have more than roughly 1000 buffers, the send fails. > > -ovidiu > > On Wed, Jun 24, 2020 at 8:54 AM Ovidiu Sas wrote: > > > > Hello Razvan, > >

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

It seems that when we have more than roughly 1000 buffers, the send fails. -ovidiu On Wed, Jun 24, 2020 at 8:54 AM Ovidiu Sas wrote: > > Hello Razvan, > > The system is a debian buster one. > I patched the code: > #ifdef IOV_MAX > LM_NOTICE("IOV_MAX=[%d]\n", IOV_MAX); > #else > LM_NOTICE

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hello Razvan, The system is a debian buster one. I patched the code: #ifdef IOV_MAX LM_NOTICE("IOV_MAX=[%d]\n", IOV_MAX); #else LM_NOTICE("no IOV_MAX\n"); #endif and I get: NOTICE:rtpengine:send_rtpe_command: no IOV_MAX in the logs. Then I patched the code again to check how many buffers

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hi, Ovidiu! I doubt this is a problem of OpenSIPS version, but rather of the OS you are running on. I suspect that error comes from the fact that the bson resulted has more than IOV_MAX elements, which if I recall correctly it was 15 on some OSes. We had a similar problem in rtpproy[1], where

Re: [OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

This is happening also on the latest 3.0. The weird thing is that opensips doesn't send anything to rtpengine. The first opensips/rtpengine exchange on the initial INVITE works ok, but the opensips/rtpengine exchange on the 200ok fails (no command is sent by opensips - confirmed by running ngrep on

[OpenSIPS-Users] rtpengine errors: can't send command to a RTP proxy (22:Invalid argument)

Hello all, I'm running opensips 3.1.0-beta (latest version) and experiencing connectivity issues to the rtpengine daemon running on the same host: ERROR:rtpengine:send_rtpe_command: can't send command to a RTP proxy (22:Invalid argument) ERROR:rtpengine:send_rtpe_command: timeout waiting reply fro

[OpenSIPS-Users] RTPEngine Setup Delay Between External Users

Hello Everyone, I'm trying to solve a problem where there is a 3-4 second delay before audio is connected when using RTPEngine between two external users. In this scenario OpenSIPS/RTPEngine is used as sort of an SBC for remote users, and there is a PBX behind OpenSIPS that just proxies the call

Re: [OpenSIPS-Users] rtpengine documentation

Actually I had read the readme and I was wondering if opensips perhaps didn't support all the flags since some were missing from the documentation. Also on the subject of DTLS I am guessing that no flags means DTLS pass through but not certain. Also on the subject of DTLS when it plays MITM i

Re: [OpenSIPS-Users] rtpengine documentation

Hello Robert, Take a look at the README file. Based on the flags, rtpengine can bridge encrypted RTP traffic to unencrypted RTP traffic. It can also do transcoding. So yes, it plays man-in-the-middle :) Regards, Ovidiu Sas On Tue, May 19, 2020 at 18:32 Robert Dyck wrote: > Perhaps someone wit

Re: [OpenSIPS-Users] rtpengine documentation

Perhaps someone with knowledge of the inner workings of rtpengine could enlighten us about the interaction between ICE and DTLS. My experience suggests that it plays man-in-the-middle and fakes the DTLS negotiation in some circumstances. Rob On Tuesday, May 19, 2020 3:15:54 P.M. PDT Giovanni M

Re: [OpenSIPS-Users] rtpengine documentation

On Tue, May 19, 2020, 20:10 Ovidiu Sas wrote: opensips rtpengine module provides a > mechanism to pass those flags as strings to the rtpengine instance. > Maybe we should add this to the documentation. > +1 +1 +1 (me, myself and I) -giovanni Regards, > Ovidiu Sas > > On Sat, May 16, 2020 at

Re: [OpenSIPS-Users] rtpengine documentation

For complete rtpengine documentation, take a look at the rtpengine README.md: https://github.com/sipwise/rtpengine There are new feature added with each new rpengine release. The opensips rtpemgine module documentation (with respect to flags) should be seen as a starting point.and the rtpengine REA

Re: [OpenSIPS-Users] rtpengine documentation

Hi Rob, I'm interested to follow your thread to hear more about this, I have found that some flags are valid yet undocumented during initial setup of some RTC compatable proxies. Two in particular: DTLS-passive and SDES-disable both of which appear to influence behaviour of RTPEngine in accordanc

[OpenSIPS-Users] rtpengine documentation

I am wanting to convert my config/script to use rtpengine instead of rtpproxy. I think it would better deal with webrtc. After looking at some examples I found, I see a couple of parameters that are not mentioned in the opensips documentation. First there is the offer/answer option ice=force-rel

Re: [OpenSIPS-Users] RTPengine Unknown flag encountered: 'codec-mask-PCMA'

Thank you Giovanni, That maked the trick. localhost rtpengine[2358]: INFO: [f17a2e414a14166a]: Enabling transcoding engine On Sat, May 2, 2020 at 2:39 AM Giovanni Maruzzelli wrote: > Those flag are only valid in "offer" > > > > On Sat, May 2, 2020 at 8:48 AM Mario San Vicente > wrote: > >> H

Re: [OpenSIPS-Users] RTPengine Unknown flag encountered: 'codec-mask-PCMA'

Those flag are only valid in "offer" On Sat, May 2, 2020 at 8:48 AM Mario San Vicente wrote: > Hello everyone, > > I have been testing transcoding and so far i can bridge the audio using > rtpengine and it works fine. But when i try to transcode with the > following lines a get a warning: > >

Re: [OpenSIPS-Users] RTPengine Unknown flag encountered: 'codec-mask-PCMA'

Mu intention is to change the incoming codec PCMA to outgoing PCMU . I have changed the flags to the following: $var(reflags) = "strip=PCMA transcode=PCMU replace-session-connection replace-origin"; Now i get not error / warning logs, but it is not changing the codec offer. Thanks On Sat, May

[OpenSIPS-Users] RTPengine Unknown flag encountered: 'codec-mask-PCMA'

Hello everyone, I have been testing transcoding and so far i can bridge the audio using rtpengine and it works fine. But when i try to transcode with the following lines a get a warning: route[rtp_answer] { $var(reflags) = "codec-mask-PCMA transcode-PCMU replace-session-connection replace-or

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

Oops... I answered early. You had more replies. I've tried it everywhere. Before, after, branch route, script route, etc. The only combination I haven't tried is moving the rtpengine_offer call to a script route from the branch route it's in now, and then doing my custom replacements in the bran

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

David, I have. The "origin" flag replaces the IP address of the origin but nothing before it. The "session-connection" digs into the various sessions that may be defined in the SDP and replaces all the c= lines (important for RTP flows), but unfortunately it doesn't do anything for the s= (the s

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

Though reading through it, it may not be what your're looking for. I think the SDP gets replaced when you call the rtpengine manage command. Have you tried doing the replace _after_ calling the manage command? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

with: https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html#rtpengine.f.rtpengine_offer 5.2. rtpengine_offer([flags]) *replace-origin* - flags that IP from the origin description (o=) should be also changed. Regards, David Villasmil email: david.villasmil.w...@gmail.com phone:

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

I believe you can use this: replace Similar to the flags list. Controls which parts of the SDP body should be rewritten. Contains zero or more of: - origin Replace the address found in the *origin* (o=) line of the SDP body. Corresponds to *rtpproxy* o flag. - session connec

[OpenSIPS-Users] rtpengine + manual SDP manipulations

Hello, I'm running OpenSIPS 2.4 and rtpengine 7.0. I have the following commands in my script: route[sanitize_sdp] { subst_body('/^o=\S+ /o=- /'); subst_body('/^s=.*/s=-\r/'); } This works fine, unless I'm also using rtpengine, in which case these subst_body() changes are lost.

Re: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL

I have to use local cache to workaround. *In INVITE statement:* $avp(setid) = 1; # group of rtp engine servers / Ex: 1 for FR / 2 for USA if(cache_store("local","$ci","$avp(setid)",1200)) { xlog("L_INFO","$ci -- Cache Store Name: $ci with port: $avp(setid)\n"); *in CANCEL statement:* if(cache_fe

Re: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL

OK thanks, I will write something about that ;) I tested your solution, it works for BYE but not force CANCEL. Indeed, for BYE from my SBC enters in loose_route statement, but not for a CANCEL. is there a simplest way to choose the good rtpengine ? I want use a group of rtpengine (like id:1) for

Re: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL

No, it only survives until the end of transaction :). I agree it would be nice to have it persistent throughout the dialog, but that's not in there yet. But you can always open a feature request for this :): https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan On 11/20/18 12:15 P

Re: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL

Hi Răzvan, thanks a lot for your answer ! Ok I will try this workaround But I don't understand the rtpengine module documentation ( http://www.opensips.org/html/docs/modules/2.4.x/rtpengine.html) If I have -> modparam("rtpengine", "setid_avp", "$avp(setid)") and I set setid_avp to 1 (for my examp

Re: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL

Hi, Mickael! AVPs are only available during transaction - initial request and its replies, but it is not avaialble to sequential requests (such as CANCEL or BYE). If you want to store a value throughout the entire dialog, you should save the variable in a dialog variable[1], something like thi

[OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL

Hi all, I have a rtpengine with opensips, when I use rtpengine without setid_avp all works like a charm modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221") *When I want use setid_avp:* modparam("rtpengine", "setid_avp", "$avp(setid)") modparam("rtpengine", "rtpengine_sock", "1 == ud

[OpenSIPS-Users] RTPEngine node selection

Hi I have a configuration with OpenSIPS 2.4.1 and an rtpengine set which contains 2 rtpengine nodes. At some points I notice that OpenSIPS will issue an rtpengine_offer to node1, and then the subsequent rtpengine_answer will be sent to node2. The rtpengine_answer call invariably fails because rt

[OpenSIPS-Users] rtpengine stats

Hello: I am using opensips 2.4.2, working with the rtpengine module trying to get stats per call leg.  I have tried using the $rtpstat(STAT)[index] function with to-tag/from-tag but it seems to only return the first called stat... with subsequent calls returning an empty string. Example

Re: [OpenSIPS-Users] rtpengine

Thanks a lot!!! На вт, 2.10.2018 г. в 10:11 ч. Răzvan Crainea написа: > Yes. You should also use the -x 1 parameter, to trigger the command only > using the callid. > > Best regards, > Răzvan > > On 10/1/18 7:22 PM, Dragomir Haralambiev wrote: > > Hello, > > > > If I use -b parameter of rtpengin

Re: [OpenSIPS-Users] rtpengine

Yes. You should also use the -x 1 parameter, to trigger the command only using the callid. Best regards, Răzvan On 10/1/18 7:22 PM, Dragomir Haralambiev wrote: Hello, If I use -b parameter of rtpengine it should have active MI_XMLRPC_NG

[OpenSIPS-Users] rtpengine

Hello, If I use -b parameter of rtpengine it should have active MI_XMLRPC_NG module in Opensips? Best regards, Dragomir ___ Users mailing list Users@lists.opensips.org http://lists.opensip

Re: [OpenSIPS-Users] RTPengine provisioning by db

gt; Best regards, > > > > > > Răzvan > > > > > > > > > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote

Re: [OpenSIPS-Users] RTPengine provisioning by db

gt; > > Hi all, > > > > > > > > > > > > >

Re: [OpenSIPS-Users] RTPengine provisioning by db

te: > > > > > > > Hi all, > > > > > > > > > > > > > > nobody has any other idea? > > >

Re: [OpenSIPS-Users] RTPengine provisioning by db

> Did you check if the file is actually there after you get the > > > > > > opensipsctl error? > > > > > > > > > > > > Be

Re: [OpenSIPS-Users] RTPengine provisioning by db

03:55 PM, Alain Bieuzent wrote: > > > Hi all, > > > > > > nobody has any other idea? > > > > > > Thanks > > > > > > *De : *Users

Re: [OpenSIPS-Users] RTPengine provisioning by db

gt; Best regards, > > Răzvan > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > Hi all, > > > > > > nobody has an

Re: [OpenSIPS-Users] RTPengine provisioning by db

> > nobody has any other idea? > > Thanks > > *De : *Users au nom de Alain Bieuzent > > *Répondre à : *OpenSIPS users mailling list > *Date : *vendredi 6 juillet 2018 à 14:06 > *À : *Pasan Meemaduma , Open

Re: [OpenSIPS-Users] RTPengine provisioning by db

in Bieuzent wrote: > Hi all, > > nobody has any other idea? > > Thanks > > *De : *Users au nom de Alain Bieuzent > > *Répondre à : *OpenSIPS users mailling list > *Date : *vendredi 6 juillet 2018 à 14:06 >

Re: [OpenSIPS-Users] RTPengine provisioning by db

re à : *OpenSIPS users mailling list *Date : *vendredi 6 juillet 2018 à 14:06 *À : *Pasan Meemaduma , OpenSIPS users mailling list *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db Hi Pasan, Thanks for the reply but yes of course I have mi_fifo module load and configure. I’m using

Re: [OpenSIPS-Users] RTPengine provisioning by db

Hi all, nobody has any other idea? Thanks De : Users au nom de Alain Bieuzent Répondre à : OpenSIPS users mailling list Date : vendredi 6 juillet 2018 à 14:06 À : Pasan Meemaduma , OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] RTPengine provisioning by db Hi Pasan

Re: [OpenSIPS-Users] RTPengine provisioning by db

mmand opensipsctl fifo ds_list provide the same error about mi_fifo.   thanks   De : Users au nom de Pasan Meemaduma via Users Répondre à : Pasan Meemaduma , OpenSIPS users mailling list Date : vendredi 6 juillet 2018 à 12:23 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-

Re: [OpenSIPS-Users] RTPengine provisioning by db

: Re: [OpenSIPS-Users] RTPengine provisioning by db Hi Alain, do you have following config lines in your opensips.cfg file which enables the fifo management interface ? loadmodule "mi_fifo.so" # - mi_fifo params - modparam("mi_fifo", "fifo_n

Re: [OpenSIPS-Users] RTPengine provisioning by db

Hi Alain, do you have following config lines in your opensips.cfg file which enables the fifo management interface ? loadmodule "mi_fifo.so" # - mi_fifo params - modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") According to the error you got it doesn't seems to exists. On

[OpenSIPS-Users] RTPengine provisioning by db

Hi all, I’m trying to provionning rtpengine via mysql db, but i can’t add a new proxy via opensipctl My parameters are : ## RTPENGINE module ##

Re: [OpenSIPS-Users] rtpengine + cluster

Hello Razvan, Thank you for info I conigured multiply rtpengines cluster right now 3 opensips nodes. And rtpengine having trouble detect available rtpengines. Start up log Jun 28 13:42:58 aitossbc03 /usr/sbin/opensips[1572]: INFO:rtpengine:rtpe_test: rtp proxy found, support for it enable

Re: [OpenSIPS-Users] rtpengine + cluster

No, you don't need to add it if you don't need it. Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 06/26/2018 04:35 PM, volga...@networklab.ca wrote: Hello Razvan, Thank you, do I need add SEQUENCE  too ? CREATE SEQUENCE public.rtpengine_id_seq    START WITH 1   

Re: [OpenSIPS-Users] rtpengine + cluster

Hello Razvan, Thank you, do I need add SEQUENCE too ? CREATE SEQUENCE public.rtpengine_id_seq START WITH 1 INCREMENT BY 1 NO MINVALUE MAXVALUE 2147483647 CACHE 1 CYCLE; volga629 On Tue, Jun 26, 2018 at 3:01 AM, Răzvan Crainea wrote: Hi, Volga! You can find the raw struct

Re: [OpenSIPS-Users] rtpengine + cluster

Hi, Volga! You can find the raw structure of the rtpengine table in scripts/postgres/rtpengine-create.sql[1]. [1] https://github.com/OpenSIPS/opensips/blob/master/scripts/postgres/rtpengine-create.sql Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On

Re: [OpenSIPS-Users] rtpengine + cluster

Hello Everyone, I tried create new database with [root@canldev01-fedora27 opensips]# opensipsdbctl create opensips01 PGSQL password for postgres: INFO: creating database opensips01 ... ERROR: role "opensips_prod" already exists WARNING: Create user in database failed, perhaps they already exist?

Re: [OpenSIPS-Users] rtpengine transcoding

an interesting fact. if i set the same codecs on both phones (ALAW on phoneA and ALAW,ULAW,G722 on phoneB), everything is OK but there is the same error in rtpengine log: rtpengine[9155]: DEBUG: [431a73fa0fb9b02a port 30026]: Error parsing RTP header: invalid header version But two-way audio

Re: [OpenSIPS-Users] rtpengine transcoding

I'll illustrate what I said. This is "phoneA ---> OpenSIPS" callleg, ALAW, played in Wireshark http://rgho.st/8B4c47jCy This is "OpenSIPS ---> phoneB" callleg, ULAW, played in Wireshark http://rgho.st/6Hs8hhQ6z Even do not know why I hear nothing in the phones. I think it's that error in RTPEn

Re: [OpenSIPS-Users] rtpengine transcoding

If to play streams from pcap using Wireshark, we hear the noise (I haven't told anything during the test call, just blowed in both microphones). And it's necessary to say that streams from both call-legs are played, using right codecs (phoneA ---> OpenSIPS and OpenSIPS ---> phoneB). But I don't

Re: [OpenSIPS-Users] rtpengine transcoding

can you share a pcap trace ? De : Users au nom de Alexey Kazantsev via Users Répondre à : Alexey Kazantsev , OpenSIPS users mailling list Date : jeudi 21 juin 2018 à 12:26 À : "users@lists.opensips.org" Objet : Re: [OpenSIPS-Users] rtpengine transcoding Hi Alain, thank y

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