HI All,
I am not sure if I understand it correctly but I thought that I could use
rtpengine to redirect media packets. My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is
Asterisk<=>softphone. But I don't want Media to flow this way.
That is, I
Hello Everyone,
What flags available on latest rtpegnine to acomplish transcoding and
codec manipulation ?
Slava.
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Hey guys,
Is this the right place to ask about rtpengine (ngcp) related issues with
kernel packet forwarding?
Thanks.
--
Andy Chen
Sr. Telephony Lead Engineer
achen@ fuze.com
--
*Confidentiality Notice: The information contained in this e-mail and any
attachments may be confidential. If you
Hello.
It's not enough to install rtpproxy/rtpengine and start it. You need to
call rtpengine functions from kamailio.cfg for every SIP packet with SDP
so media will be routed "thru Kamailio".
To make it easier, I recommend to begin with rtpproxy, not rtpengine.
Default kamailio.cfg contains
in the NAT router.”
Obviously, this is not correct.
From: sr-users On Behalf Of Yu Boot
Sent: Tuesday, August 14, 2018 2:12 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Rtpengine?
Hello.
It's not enough to install rtpproxy/rtpengine and start it. You need to call
rtpengine f
).
Do I need to add rtpproxy_manage() at every SIP message?
Thank you,
-Steve
From: sr-users On Behalf Of Yu Boot
Sent: Tuesday, August 14, 2018 2:12 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Rtpengine?
Hello.
It's not enough to install rtpproxy/rtpengine and start it. You
Hello all,
I am still trying to get RTPEngine to work. At this point, when I make a call,
I do see the offer in the rtpengine log, I also see the update of the SDP (c)
changing it to 20.20.20.20. However, after that, there are no logs other than a
bunch of "timer run time = 0.nnn sec", and
Hello All,
I have an issue where Kamailio-RTPEngine-Asterisk calls work good when a
softphone UAC on an IOS phone makes an Inbound call to a WebRTC client.
However, if the softphone UAC is on Windows, it does not work (No Audio/Video).
I noticed in the Wireshark trace that when using IOS I se
Greetings.
I'm trying to set up annoncement playback for both parts of the call. I
have to play the file after the connection is established. Stop_media (all)
must be executed before playing
It turns out the construction:
block_media ("all");
play_media ("all file = / etc / kamailio / sounds
I have gathered that RTPEngine has recently, or perhaps some time ago,
evolved a recording feature set:
https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.start_recording
Does anyone have any experience using it with Kamailio? How does it
work? Any gotchas or pitfalls?
It
Hi List,
I just read on the RTPEngine page the Proxy RTP now support transcoding:
https://github.com/sipwise/rtpengine#transcoding
is there a plan to support it?
Regards
--
---
I'm SoCIaL, MayBe
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Hello!
https://github.com/sipwise/rtpengine/blob/master/README.md
You can use options like codec-transcode-... (dash instead of equals).
2018-04-09 1:40 GMT+02:00 :
> Hello Everyone,
> What flags available on latest rtpegnine to acomplish transcoding and
> codec manipulation ?
>
> Slava.
>
>
>
Hello Denys,
I am trying remove all codec and offer only G729A.
Based on doc I can do some think like this ?
codec-offer-G729
Slava
On Mon, Apr 9, 2018 at 5:05 AM, Denys Pozniak
wrote:
Hello!
https://github.com/sipwise/rtpengine/blob/master/README.md
You can use options like codec-transcod
I think you need to mask all codecs from A side and transcode g729, like
this:
*rtpengine_offer("codec-mask-all codec-transcode-G729")*
2018-04-09 12:35 GMT+02:00 :
> Hello Denys,
> I am trying remove all codec and offer only G729A.
> Based on doc I can do some think like this ?
>
> codec-offe
But if side A has G729, probably you can avoid transcoding by stripping
all codecs and offer needed.
*rtpengine_offer("codec-strip-all codec-offer-G729")*
2018-04-09 12:47 GMT+02:00 Denys Pozniak :
> I think you need to mask all codecs from A side and transcode g729, like
> this:
>
> *rtpengin
Hello Denys,
Based on what I see G729 transcoding is not supported. What would be
the options in mix with codec-strip-all and leave only g729
Slava.
On Mon, Apr 9, 2018 at 7:58 AM, Denys Pozniak
wrote:
But if side A has G729, probably you can avoid transcoding by
stripping all codecs and
Hello!
Looks like you do not have bgc729.
2018-04-10 23:57 GMT+02:00 :
> Hello Denys,
> Based on what I see G729 transcoding is not supported. What would be the
> options in mix with codec-strip-all and leave only g729
>
> Slava.
>
>
> On Mon, Apr 9, 2018 at 7:58 AM, Denys Pozniak
> wrote:
>
>
Hello Denys,
Yes it was mssing. I jsut installed and will test again.
[root@prx00 ~]# rpm -qa | grep bcg
bcg729-1.0.4-2.fc27.x86_64
Slava
On Wed, Apr 11, 2018 at 4:52 AM, Denys Pozniak
wrote:
Hello!
Looks like you do not have bgc729.
2018-04-10 23:57 GMT+02:00 :
Hello Denys,
Based on what
Hi,
When using a single RTPEngine, or a single RTPEngine per set, it is
possible to make stateless RTPEngine calls from Kamailio, since
everything is keyed by Call-ID + tag.
In other words, one can set up a call, answer it, restart Kamailio, and
still send a 'delete' to RTPEngine successfull
There is an RTPEngine mailing list, I believe, but RTPEngine questions
are often posed here given its close association with Kamailio. What's
going on?
On 9/11/20 2:57 PM, Andrew Chen wrote:
Hey guys,
Is this the right place to ask about rtpengine (ngcp) related issues
with kernel packet for
On 11/09/2020 15.29, Andrew Chen wrote:
Thanks Alex.
So it turns out my rtpengine stopped working after our latest kernel
upgrade to:
Linux sjomainrtpe30 5.3.0-1035-aws #37-Ubuntu SMP Sun Sep 6 01:17:09
UTC 2020 x86_64 x86_64 x86_64 GNU/Linux
at the time, I was running an older version 8.0
Related to tainted kernel, I faced the same issue when I deployed
rtpengine on a Suse Enterprise many months ago, so I do not really
remember the exact steps, but there is a way to disable the check of
signed kernel modules (iirc, these are only the ones coming from the
kernel source tree, so if yo
Thanks Alex.
So it turns out my rtpengine stopped working after our latest kernel
upgrade to:
Linux sjomainrtpe30 5.3.0-1035-aws #37-Ubuntu SMP Sun Sep 6 01:17:09 UTC
2020 x86_64 x86_64 x86_64 GNU/Linux
at the time, I was running an older version 8.0.x so I recompiled all the
ngcp packages under
Sorry let me clarify this line here:
"...at the time, I was running an older version 8.0.x so I recompiled all
the ngcp packages under this kernel and completed the installation without
issues.."
8.0.x is the older ngcp version. I recompiled version 9.0.1.0 under that
new kernel version 5.3.0-103
Thanks Daniel. Let me look at that option.
On Fri, Sep 11, 2020 at 3:52 PM Daniel-Constantin Mierla
wrote:
> Related to tainted kernel, I faced the same issue when I deployed
> rtpengine on a Suse Enterprise many months ago, so I do not really remember
> the exact steps, but there is a way to d
I have testes build on 5.7 kernel on CentOS 8 (custom rpm package)
You will find commit here
https://github.com/sipwise/rtpengine/issues/975
Sergey
On Fri, Sep 11, 2020 at 10:53 PM Daniel-Constantin Mierla
wrote:
> Related to tainted kernel, I faced the same issue when I deployed
> rtpengine o
On 14/09/2020 13.14, Andrew Chen wrote:
Btw Richard Fuchs, to follow up on your comment, we have a load
generator running sipp which is non-SRTP traffic.
As for the fallback, how does that work exactly? We tried the
following today and it seems to have helped:
- Removed "--table" startup para
Btw Richard Fuchs, to follow up on your comment, we have a load
generator running sipp which is non-SRTP traffic.
As for the fallback, how does that work exactly? We tried the following
today and it seems to have helped:
- Removed "--table" startup param in systems file
- Uncommented "no-fallback
Sergey Safarov,
Thanks for the response but we're not a CentOS house. Just Ubuntu.18.04.
One other question. Is Daniel's suggestion the only way to get around
kernel forwarding with our current setup? Will there be support for it in
the future without disabling module signature check?
On Mon,
Hmm...interesting.
Is there logging which states the fallback is issued? If so, can someone
share?
Thanks.
On Mon, Sep 14, 2020 at 1:28 PM Richard Fuchs wrote:
> On 14/09/2020 13.14, Andrew Chen wrote:
> > Btw Richard Fuchs, to follow up on your comment, we have a load
> > generator running s
Hi,
I have a question about RTPEngine set IDs:
As of Kamailio 5.x, it appears the RTPEngine module has got quite smart. Even if
I don't set `setid_avp`, it still knows which RTPEngine instance and set
ID was engaged by an initial rtpengine_offer() / rtpengine_manage()
command. This knowledge pers
Hi there,
Well it's mostly rtpengine question but didn't know where should I send
it and probably the answer will be more or less useful for kamailio users.
By default if an offer has ICE candidates rtpengine leaves them
untouched and just inserts itself as an additional low priority
candidat
Hi,
I want to implement selective transcoding, e.g. avoiding Transcoding
between a HD codec (G722) and a non HD Codec (G711).
In case the caller does offer HD codecs, this is fine, e.g.
if(sdp_with_codecs_by_name("G722,OPUS")) {
rtpengine_offer("codec-transcode=OPUS codec-transcode=G722
codec-
rtpengine module README tells this about module dependencies:
3.1. Kamailio Modules
The following modules must be loaded before this module:
tm module - (optional) if you want to have rtpengine_manage() fully
functional
Looks like this is not correct. If Kamailio connects to rtpengi
I try to workout if - currently it would work, or - where and how to debug
more:
I face - 2 interfacec - public internet (so, TLS + sRTP) is desired
and private - old infrastructure - i mus only use plain RTP
172.23.9.70 - private ip - from this endpoint of kamailio and rtpengine should
send
Hi guys,
I'm using rtpengine (compiled from source) and I see a lot of difference in
cpu usage comparing with rtpproxy. Is this normal?
With 600 calls:
cpu
*41.7%* rtpengine
#rtpengine threads
4
# lscpu
Architecture: x86_64
CPU op-mode(s):32-bit, 64-bit
Byte Order:
Hi Alex,
I had the opportunity to play around with this recording and it works very
well. We have it implemented on production by the way.
The information about how it works, recording formats you can find in the
README here (https://github.com/sipwise/rtpengine).
Once you have the recording you
On Thu, Aug 10, 2017 at 11:40:28AM -0400, Alberto Llamas wrote:
> Once you have the recording you need a third-party tool to convert those
> files to a wav format for instance. It depends on the codec the call is
> being recorded the convert tool you have to use like ffmpeg.
Thank you for your fe
On 10/08/17 11:40 AM, Alberto Llamas wrote:
Hi Alex,
I had the opportunity to play around with this recording and it works
very well. We have it implemented on production by the way.
The information about how it works, recording formats you can find in
the README here (https://github.com/sip
On Thu, Aug 10, 2017 at 12:22:32PM -0400, Richard Fuchs wrote:
> There's actually two distinct recording methods implemented. The one you're
> describing simply outputs pcap files containing the raw packets involved in
> the call. The other method involves the aforementioned external recording
> d
On 10/08/17 12:23 PM, Alex Balashov wrote:
On Thu, Aug 10, 2017 at 12:22:32PM -0400, Richard Fuchs wrote:
There's actually two distinct recording methods implemented. The one you're
describing simply outputs pcap files containing the raw packets involved in
the call. The other method involves t
I have succeeded in prototyping a recording setup using the 'proc'
method.
However, I've got one issue I can't seem to figure out. On inbound calls
only, the inbound (caller) leg on the PSTN side seems to show up
interleaved/stuttered in the recording, and also slowed down
considerably. This does
Also, the "proc" recording method has an interesting pipeline, involving
an ephemeral metadata file and a memory sink exposed through /proc, and
a userspace daemon picking up the data and writing it to disk (if this
doesn't happen, audio frames going into the sink are discarded).
This pipeline le
On 12/08/17 03:49 PM, Alex Balashov wrote:
Also, the "proc" recording method has an interesting pipeline, involving
an ephemeral metadata file and a memory sink exposed through /proc, and
a userspace daemon picking up the data and writing it to disk (if this
doesn't happen, audio frames going int
Hello,
wondering if anyone has a quick answer to spare some time searching the
web or source code -- is there a way to offer optional SRTP (just SDES
is also fine) in SDP with RTPEngine? The use case is when the target
device is not yet known to support (or not) SRTP.
There are couple of ways in
On 2018-02-13 03:34 PM, Social Boh wrote:
Hi List,
I just read on the RTPEngine page the Proxy RTP now support transcoding:
https://github.com/sipwise/rtpengine#transcoding
is there a plan to support it?
Kamailio's rtpengine module can already make full use of rtpengine's
transcoding capabi
Thank you for your answer but I don't see any option to activate
transcoding on Kamailio RTPEngine module
Regards
---
I'm SoCIaL, MayBe
El 13/02/2018 a las 17:49, Richard Fuchs escribió:
> On 2018-02-13 03:34 PM, Social Boh wrote:
>> Hi List,
>>
>> I just read on the RTPEngine page the Proxy RTP
On 2018-02-13 06:07 PM, Social Boh wrote:
Thank you for your answer but I don't see any option to activate
transcoding on Kamailio RTPEngine module
Just use the options as described in rtpengine's readme. Kamailio's
module will pass them through just fine.
Cheerse
The attributes passed to rtpengine from the offer/answer/manage/query functions
are not all documented in the Kamailio RTPengine module docs, but all values
are accepted.
-- Alex
--
Sent via mobile, please forgive typos and brevity.
___
Kamailio (S
Hello! Good news!
Please clarify how recording (pcap and proc) will work in case of
transcoding if there is an influence on each other.
On 14 February 2018 at 00:19, Alex Balashov
wrote:
> The attributes passed to rtpengine from the offer/answer/manage/query
> functions are not all documented in
On 2018-02-14 05:40 AM, Denys Pozniak wrote:
Hello! Good news!
Please clarify how recording (pcap and proc) will work in case of
transcoding if there is an influence on each other.
They're completely independent of each other. The source data for
recording are the packets as they're received
So if we have 2 legs: opus (webrtc side) and pcm (sip side), pcap file will
have also 2 incoming RTP streams: opus and pcm, correct?
On 14 February 2018 at 14:19, Richard Fuchs wrote:
> On 2018-02-14 05:40 AM, Denys Pozniak wrote:
>
>> Hello! Good news!
>> Please clarify how recording (pcap and
On 2018-02-14 08:55 AM, Denys Pozniak wrote:
So if we have 2 legs: opus (webrtc side) and pcm (sip side), pcap file
will have also 2 incoming RTP streams: opus and pcm, correct?
Yes, correct
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Hello,
So which version is the one with transcoding and repacketization support
enabled? I don't see anything relevant in debian/changelog, not even in
current master branch (which apparently will become version 6.2.x.x at some
point). Is this baked in latest release mr6.1.1.1? Thanks!
BR,
George
On 2018-02-14 04:07 PM, George Diamantopoulos wrote:
Hello,
So which version is the one with transcoding and repacketization
support enabled? I don't see anything relevant in debian/changelog,
not even in current master branch (which apparently will become
version 6.2.x.x at some point). Is t
Hello,
I'm trying to use this configuration:
Softphone A audio codec g722 and Opus
Softphone B audio codec PCMA and PCMU
Kamailio with rtpengine_manage with codec-transcode-PCMU
codec-transcode-PCMA in the request
Softphone A call Softphone B
The RTPEngine DEBUG show many Decoder Error:
http
On 2018-02-15 07:35 AM, Social Boh wrote:
Hello,
I'm trying to use this configuration:
Softphone A audio codec g722 and Opus
Softphone B audio codec PCMA and PCMU
Kamailio with rtpengine_manage with codec-transcode-PCMU
codec-transcode-PCMA in the request
Softphone A call Softphone B
The RT
I realize the transcoding support is rather new, however has anyone run any
real world tests on it yet? Trying to get a sense of what the performance will
be.
I'm aware that it's forced into userspace for transcoding so it's not *nearly*
as efficient as the iptables module, however even if I h
Greetings list.
This might not be related to Kamailio but I thought someone would be able
to give a hand in resolving dependency during the installation of rtpengine
on Debian 8 Jessie.
It is complaining about below dependencies.
root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage
d
Hello,
In RTPEngine recording-daemon, I enabled "forward-to" option to
unix_socket. All things work right, I got metadata at first, then
media packets (RTP-UDP-IP).
The metadata for each calls is received just at first packet, but i
want to know, if i have more than one concurrent active call in
RT
Hello all,
I'm currently doing some experiments with RTPENGINE and I need some help
configuring it.
First scenario i'm under NAT and Kamailio receives an INVITE with a private
address, rewrites (c) line on SDP with it's own IP address and relays the
message to the destination endpoint. After that
I found this from 2015...
https://github.com/kamailio/kamailio/pull/390
If it's still holding true, looks like it's rehashed?
--fred
On Mon, 2020-05-25 at 18:02 -0400, Alex Balashov wrote:
> Hi,
>
> When using a single RTPEngine, or a single RTPEngine per set, it is
> possible to make state
hi
i m using kamailio 5.3 ans install rtpengine on different server
kindly help me what will be the configuration in case of kamailio
and rtpengine
thank you
--
*Regards:*
Gaurav Kumar
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Hi,
Can anyone give a quick answer regarding RtpEngine and support for iLBC
codec?
I have followed the instructions how to build and enable codecs according
to the GitHub manual. But iLBC codec is missing when performing "rtpengine
--codecs".
I have tried to google around for iLBC references, bu
Hi!
Happy new year to all!!!
Look like I am first in this year wit hthe questions in this list :-).
I'm using stateless kamailio and RTPengnine to build some kind of the
stateless cluster
I found that kamailio keeps some data in the SHMEM in case of using
RTPengine module even if it is not a rtp
On Thu, Mar 14, 2019 at 02:47:24PM +0200, Vitalii Aleksandrov wrote:
> Well it's mostly rtpengine question but didn't know where should I send it
> and probably the answer will be more or less useful for kamailio users.
I get the feeling. You could open an issue in github, but
sipwise/rtpengine de
On 14.03.19 17:09, Daniel Tryba wrote:
On Thu, Mar 14, 2019 at 02:47:24PM +0200, Vitalii Aleksandrov wrote:
Well it's mostly rtpengine question but didn't know where should I send it
and probably the answer will be more or less useful for kamailio users.
I get the feeling. You could open an is
On Thu, Mar 14, 2019 at 06:01:41PM +0200, Vitalii Aleksandrov wrote:
> > What is wrong with the default behavior? That adds ICE records and
> > rewrites SDP c=.
> When a call goes through multiple proxies and every proxy inserts itself SDP
> becomes really huge. What I like in "force-relay" is that
On Thu, Mar 14, 2019 at 06:01:41PM +0200, Vitalii Aleksandrov wrote:
What is wrong with the default behavior? That adds ICE records and
rewrites SDP c=.
When a call goes through multiple proxies and every proxy inserts itself SDP
becomes really huge. What I like in "force-relay" is that it rem
On Fri, Mar 15, 2019 at 12:37:16PM +0200, Vitalii Aleksandrov wrote:
> Oh, it actually does. If you use ICE=force, rtpengine removes all ICE
> candidates and inserts its own and both call participant can't to talk to
> each other directly but still can use ICE to establish media streams to
> rtpeng
On 15/03/2019 06.37, Vitalii Aleksandrov wrote:
On Thu, Mar 14, 2019 at 06:01:41PM +0200, Vitalii Aleksandrov wrote:
What is wrong with the default behavior? That adds ICE records and
rewrites SDP c=.
When a call goes through multiple proxies and every proxy inserts
itself SDP
becomes really
On 21/05/2019 11.48, Carsten Bock wrote:
Hi,
I want to implement selective transcoding, e.g. avoiding Transcoding
between a HD codec (G722) and a non HD Codec (G711).
In case the caller does offer HD codecs, this is fine, e.g.
if(sdp_with_codecs_by_name("G722,OPUS")) {
rtpengine_offer("cod
Hi Richard,
Thanks for your reply, understood.
The issue is following (not really an issue, more an optimization):
At the moment of the offer, I simply don't know, what will be supported by
the callee. G711 is however always supported, that's why I don't want to
strip it. I send the call to our u
On 21/05/2019 18.42, Carsten Bock wrote:
Hi Richard,
Thanks for your reply, understood.
The issue is following (not really an issue, more an optimization):
At the moment of the offer, I simply don't know, what will be
supported by the callee. G711 is however always supported, that's why
I don
Hi,
the problem is, the device (a VoLTE phone) will ALWAYS offer AMR-WB and
AMR-NB and we will need to do transcoding to G722 / G711 in most cases
(unless the B-Party actually offers AMR-WB/AMR-NB).
The Answer may be pure G711 (RTPEngine will do Transcoding G711 <=> AMR-WB
[meaningless]) or G722 (
On 29/05/2019 03.25, Carsten Bock wrote:
Hi,
the problem is, the device (a VoLTE phone) will ALWAYS offer AMR-WB
and AMR-NB and we will need to do transcoding to G722 / G711 in most
cases (unless the B-Party actually offers AMR-WB/AMR-NB).
The Answer may be pure G711 (RTPEngine will do Transco
Hi!
Wanted to clarify some info about rtpengine working within HA-environment.
Idea is we have multiple rtpengines under same setid.
So, when I'm calling rtpengine_offer() I really don't care which is selected.
But than might happen situation, where this call(dialog) is moved to other
Kamailio
Hi,
Sorry if I missed any developments on this, but:
Kamailio does keep a runtime association of Call-ID+Ftag => RTPEngine instance,
which is significant if an RTPEngine set with multiple members is used, or if a
non-default set is used for a given call.
When Kamailio is restarted, this runtim
To: SR-Users
Subject: [SR-Users] RTPengine in Debian
HI all,
Just a quick heads up that RTPengine is now available in Debian sid/unstable
mainline repositories. Big thanks to our own Victor Seva and Guillem Jover for
making this possible.
Cheers
Hello,
good news, indeed! Thanks to all involved!
Not being very familiar with all Debian releasing policies, does that
mean the next Debian stable (v 12) will ship it? Or is it going to take
one more release cycle, to go from Debian Unstable to Testing, then to
future Stable (v 13)?
Cheers,
Da
Hi,
On 6/15/22 11:46, Daniel-Constantin Mierla wrote:
Hello,
good news, indeed! Thanks to all involved!
Not being very familiar with all Debian releasing policies, does that
mean the next Debian stable (v 12) will ship it? Or is it going to take
one more release cycle, to go from Debian Unsta
Clear now, thanks!
Cheers,
Daniel
On 16.06.22 20:41, Victor Seva wrote:
> Hi,
>
> On 6/15/22 11:46, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> good news, indeed! Thanks to all involved!
>>
>> Not being very familiar with all Debian releasing policies, does that
>> mean the next Debian stable
Hi All,
Hope you are all well?
I am hoping you can help with a bit of syntax I am struggling to get working in
the intended way. (We are using Kamailio 5.5.3 and using Kemi/python.)
The scenario is in AWS where we have separate Kamailio and RTPengine instances,
with a cluster of FreeSWITCH EC2
On Wed, May 03, 2017 at 12:10:57PM -0300, Diego Nadares wrote:
> I'm using rtpengine (compiled from source) and I see a lot of difference in
> cpu usage comparing with rtpproxy. Is this normal?
Yes, rtpengine uses less cpu (when configured correctly)
> With 600 calls:
>
> cpu
> *41.7%* rtpengin
On 05/03/2017 11:10 AM, Diego Nadares wrote:
Hi guys,
I'm using rtpengine (compiled from source) and I see a lot of
difference in cpu usage comparing with rtpproxy. Is this normal?
With 600 calls:
cpu
*41.7%* rtpengine
System CPU or User CPU?
If kernel forwarding is not in use, I would st
Thanks for your replies.
>System CPU or User CPU?
Well, it's cpu from top.
%Cpu0 : 8.3 us, 6.2 sy, 0.0 ni, 84.4 id, 0.0 wa, 0.0 hi, 1.0 si,
0.0 st
%Cpu1 : 6.2 us, 5.2 sy, 0.0 ni, 86.5 id, 0.0 wa, 0.0 hi, 2.1 si,
0.0 st
PID USER PR NIVIRTRESSHR S %CPU %MEM TI
Hi
I'm looking to check if what I'm trying to configure is feasible .
UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN
192.X.X.131)
I have a WN IP forwarded to and internal Kamailio server which is sitting in
front of an asterisk server. SIP requests are sent to aste
Hi!
I’m trying to make RTPEngine to work with calls Browser <-> SIP.
Call From SIP to Browser working perfectly, but when calling to SIP from
Browser, on 200 OK (sdp) I don’t receive
...
a=rtcp-mux
...
lines, but it was told to RTPEngine on reply_route to add this. And this line
is mandatory for
On 09/05/2017 02:32 AM, Daniel-Constantin Mierla wrote:
Hello,
wondering if anyone has a quick answer to spare some time searching the
web or source code -- is there a way to offer optional SRTP (just SDES
is also fine) in SDP with RTPEngine? The use case is when the target
device is not yet kno
On 05.09.17 16:08, Richard Fuchs wrote:
> On 09/05/2017 02:32 AM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> wondering if anyone has a quick answer to spare some time searching the
>> web or source code -- is there a way to offer optional SRTP (just SDES
>> is also fine) in SDP with RTPEngin
Hello,
On Tue, Sep 5, 2017 at 4:08 PM, Richard Fuchs wrote:
> That's not currently supported (neither as an offerer nor as an accepter).
> AFAIK the usual mantra is to offer SRTP first and then fallback to RTP when
> a "not supported" (415) is received.
I tried the solution in a free minute toda
Hello,
before I'm using 2.6.8 FFMPEG version
Installing 2.8.13 from rpmfusion repositories (CentOS 7) the problem was
solved.
Thank you.
Regards
---
I'm SoCIaL, MayBe
El 15/02/2018 a las 07:50, Richard Fuchs escribió:
> On 2018-02-15 07:35 AM, Social Boh wrote:
>> Hello,
>>
>> I'm trying to u
Hi,
Quick (and simple) question:
Can someone provide a working RTPengine request, to enable Transcoding?
I know, this is possible, but I thought before looking into deep, how this
should be configured for the rtpengine_offer() command, could someone
provide an example?
Thanks,
Carsten
_
Hello,
Is there a study comparing the performance (quality and cpu/ram consumption)
of codecs transcoding between rtpengine and asterisk/freeswith?
Is rtpengine more efficient?
Regards
Abdoul
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamai
Hi,
We've seen around 400 concurrent calls for G711<->Opus on a rather slow
12-core E5-2420@1.90GHz for some first preliminary benchmarks.
Andreas
On 2018-02-26 21:27, Brooks Bridges wrote:
> I realize the transcoding support is rather new, however has anyone run
> any real world tests on it yet
| web: www.o1.com
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Andreas Granig
Sent: Tuesday, February 27, 2018 3:12 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] rtpengine transcoding benchmarks?
Hi,
We've seen aroun
Andreas-
> We've seen around 400 concurrent calls for G711<->Opus on a rather
> slow 12-core E5-2420@1.90GHz for some first preliminary benchmarks.
Was that using narrowband (8 kHz) for Opus ? Otherwise sampling rate
conversion is needed. Also I assume Opus DTX was disabled.
Also at what Opus
On 2018-03-01 01:40 PM, Jeff Brower wrote:
Andreas-
We've seen around 400 concurrent calls for G711<->Opus on a rather
slow 12-core E5-2420@1.90GHz for some first preliminary benchmarks.
Was that using narrowband (8 kHz) for Opus ? Otherwise sampling rate
conversion is needed. Also I assume
> On 2018-03-01 01:40 PM, Jeff Brower wrote:
>> Andreas-
>>
>>> We've seen around 400 concurrent calls for G711<->Opus on a rather
>>> slow 12-core E5-2420@1.90GHz for some first preliminary benchmarks.
>> Was that using narrowband (8 kHz) for Opus ? Otherwise sampling rate
>> conversion is neede
On 04/06/2018 11:31 AM, Aqs Younas wrote:
Greetings list.
This might not be related to Kamailio but I thought someone would be
able to give a hand in resolving dependency during the installation of
rtpengine on Debian 8 Jessie.
It is complaining about below dependencies.
root@debian-769mb-m
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