On 24/05/2024 15.34, Jeff Brower wrote:
Hi Richard-
Thanks very much for your reply. Please allow me to answer in reverse
order - first in the Wireshark screencap (see link below) the two red
arrows are pointing at media packets, not SID. The sequence number
advances by 1 but the RTP
I don't know if my original reply to this went through to the list or
not, so I'm sending it again:
Is this stream actually produced by rtpengine (i.e. from transcoding) or
just a relay? Because this looks like just regular DTX behaviour from a
normal AMR sender, i.e. send a SID frame and
On 10/05/2024 04.49, palany wrote:
Thank you Richard. I have changed my rtpengine.conf as per your
instruction and now sdp m is now showing the correct ip but the o is
still showing the private ip as below.
The o= line is not relevant for the media flow, but you can add
`replace-origin` to
On 09/05/2024 10.16, palany via sr-users wrote:
I have an AWS ec2 instance with Kamailio and rtpengine configured. I
am having a challenge of no voice on calls. I have sip client on my
mobile phone and soft phone on my laptop and the two are on different
networks behind nat. Signaling is
On 10/04/2024 17.46, David Cunningham wrote:
I was asking whether OpenSSL was used because of a question we had
about FIPS validation. FIPS requires that all cryptography components
go through a validation process, which some versions of OpenSSL (but
not all) have done.
My understanding is
On 09/04/2024 23.14, Alex Balashov via sr-users wrote:
Exchanging keys directly in the SDP body is rather suboptimal from a security
standpoint, even if the signalling is encrypted, but it's certainly simpler.
I suppose that makes DTLS "more secure", but in every other sense, I'm not sure DTLS
On 09/04/2024 17.40, David Cunningham via sr-users wrote:
How does rtpengine get the TLS certificates, and what crypto library
does it use (openssl?).
SRTP itself doesn't use any certificates, and is not TLS. The underlying
cipher (AES) is provided by OpenSSL, while the SRTP implementation
On 15/12/2023 14.04, [EXT] Calvin E. via sr-users wrote:
Probably this:
https://github.com/sipwise/rtpengine/blob/master/debian/changelog
ngcp-rtpengine (12.1.0.0+0~mr12.1.0.0) unstable; urgency=medium
* [2fa121c] MT#54294 add GPU support
* [b3544be] MT#54294 packaging for rtpengine-gpu
On 08/11/2023 15.33, [EXT] David Cunningham via sr-users wrote:
Hello,
We have a Kamailio configuration with the following, however all the
load is going to the server at 22.22.22.22 (hiding the real IP
obviously) instead of being shared evenly between 22.22.22.22 and
33.33.33.33. Can anyone
On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:
Hi,
did you pass over the ice stuff from webrtc to the udp side? You could
strip that of with rtpengine options.
With recent versions we even have explicit SDP manipulations options, so
you can use rtpengine to strip attributes
On 12/10/2023 09.02, [EXT] Leonardo Arena via sr-users wrote:
I've opened the captured traffic with Wireshark and I've found out
that Kamailio is actually forwarding the call with a double SDP
header. ngrep was showing only a partial header. I think this happens
because when forwarding the call
On 26/09/2023 08.10, [EXT] Duarte Rocha via sr-users wrote:
Hello all,
I have my rtpengine configured to use ports 2 - 5 for RTP in
rtpengine.conf.
Now i want to able to select the port or the port range in a call by
call basis. Is that possible?
Not currently possible with
On 11/05/2023 16.00, [EXT] Calvin E. wrote:
We added a listen on localhost and forced the outside application
server to return 127.0.0.1 instead of the IP of Kamailio. This worked
as expected, but wasn't the solution we were looking for.
It turns out that enabling sysctl net.ipv4.ip_forward=1
On 02/05/2023 12.13, [EXT] Benoît Panizzon wrote:
Invite+SDP from B to A
It looks like this works as expected. A SDP c= line is created
containing an ipv6 address.
Sorry, I was mistaking. No it is not :-/
So I guess I have to look at the location or RURI-Host to determine
the IP protocol and
On 27/04/2023 11.08, [EXT] Kiss Zoltán wrote:
Everything is working fine, but with some clients (like Grandstream
phone) the RTCP session wants to go tot he private address of the
phone. Here is the log of one of these strange calls:
Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO:
On 25/04/2023 10.59, [EXT] Olle E. Johansson wrote:
One workaround (kludge/hack) I've seen is to store the last successful SDP
somewhere, then intercept and drop the 488 and replay the last offer/answer
with the stored SDPs. (Absolutely not something I would consider a proper
solution…)
That
On 25/04/2023 02.39, [EXT] Olle E. Johansson wrote:
So how do we handle a reinvite that is not accepted and tell RTPengine to fall
back?
Consider:
1. Successfull INVITE/200OK/ACK with alaw only
- RTPengine setup and active
2. Re-invite with video added
- RTP engine active with video
On 17/04/2023 02.55, [EXT] Tim Bowyer wrote:
Hello Everyone,
Looks to be related to the team interface after all, which is strange.
Have confirmed that after getting rid of the LACP/Team interface,
media is flowing as expected after kernelization.
...
Is it worth opening an issue in the
On 22/03/2023 08.19, [EXT] Tim Bowyer wrote:
Evening!
I ditched firewalld and swapped to configuring iptables manually…
I’ve also made some basic calls with media going in/out of the same
interface and I’m still seeing the audio stop completely or become
one-way once kernelized.
On the
This is still the same thing that I already responded to on GH, as well
earlier on this mailing list:
https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio.org/message/WG44P2L4X5SLUG6HUJ3QQDI7R6G2RGPT/
You have no connectivity to your calling party. No ICE candidates are
As mentioned on GH, you have 1) rtpengine running on a private IP
address, so is not directly reachable from the public internet, and 2)
an offer from a WebRTC client using trickle ICE, which doesn't provide
any ICE candidates, and so that client isn't reachable from rtpengine.
Either give
On 02/03/2023 09.51, [EXT] Benoit Panizzon wrote:
onreply_route[MANAGE_REPLY]
{
[...]
rtpengine_manage();
xlog("L_INFO", "$cfg(route): $rm reply MOS:
$avp(mos_average)\n");
}
Messages pass those blocks, $avp(mos_average) is 'null' no mater what.
What am I
On 02/03/2023 22.13, [EXT] Tim Bowyer wrote:
I’m having the same issue but believe it’s related to my network topology.
I have multiple carrier-facing NIC’s and an internal NIC on each media
proxy.
Is this configuration supported?
This should work fine as long as it's "just" normal IP
On 24/01/2023 06.05, [EXT] John Hardiman wrote:
...
So anything between FreeSWITCH and Kamailio/RTPengine uses the private
IP's for media, the rest use public IP's.
It is good to note that the RTPengine's are the same in each set (3
total RTPengine's), just using the public/private IP's
On 20/01/2023 10.10, [EXT] Krzysztof Drewicz wrote:
I think that i need more in depth understanding how this works - i
need to use rtpengine_ofer - before TLS w/ sRTP hits my kamailio
script ? and then, rtpengine_manage towards the no-SRTP endpoint, and
then - finally rtpengine_answer - ? So - 6
On 20/01/2023 01.19, kdrewicz+kamai...@cludo.pl wrote:
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
...
all i do is:
if
On 16/01/2023 14.48, [EXT] Mohammad Reza Keshavarzianpoor wrote:
What is the ethernet network speed?
its 1Gb esxi
Try outside of a virtualised environment.
Cheers
__
Kamailio - Users Mailing List - Non Commercial Discussions
To
On 16/01/2023 11.25, [EXT] Mohammad Reza Keshavarzianpoor wrote:
yes I tried tcp dump and it says ubuntu is dropping packets
618916 packets captured
1242894 packets received by filter
623578 packets dropped by kernel
"Packets dropped by kernel" means that some aspect of the system isn't
On 13/12/2022 09.04, [EXT] Michel Pelletier wrote:
Hello,
Thank you for your reply. I checked the versions and they look good.
xt_RTPENGINE is 11.1.1.3 and the daemon is 11.1.1.3-1~bpo11+1.
Looking at /proc/rtpengine/0/list I see the packet and byte counters
incrementing normally with 0
On 12/12/2022 14.53, [EXT] Michel Pelletier wrote:
I am proxying all RTP through RTPEngine. Everything works fine until
about 5 seconds into the call, when rtpengine enters kernelization,
after which all RTP forwarding ceases. I've checked the required
iptables entries, and all looks good.
On 14/10/2022 07.22, [EXT] Gerry Kernan wrote:
Hi Henning
Issue could be that I’m trying to do something that’s not possible .
We have an asterisk server behind kamailio . kamailio and asterisk
are on the same subnet 10.3.1.0/24 . kamailo/rtpeengine are configured
to advertise its WAN IP .
On 10/10/2022 08.58, [EXT] Gerry Kernan wrote:
Hi
Im using rtpengine_offer as below
rtpengine_offer("trust-address replace-origin
replace-session-connection ICE=remove")
is there another flag I need to set to change c=in to the WAN alias?
rtpeengine.conf
interface as set as below
###
On 09/10/2022 10.52, [EXT] Stepan Kislyakov wrote:
Hi,
I'm using latest kamailio 5.6.2 and rtpengine.
I need to play announcement to both parties when call is answered by
callee. Now I run play_media in event route but when audio start
announcement sounds like we have a 50% packet loss.
On 04/10/2022 10.59, [EXT] palany wrote:
Hi Richard
I installed from the package from git (git clone
https://github.com/sipwise/rtpengine.git) and built the .deb packages and
use the following command to install the packages.
dpkg -i ngcp-rtpengine-daemon_*.deb ngcp-rtpengine-iptables_*.deb
On 04/10/2022 08.24, [EXT] palany wrote:
I have installed RTPENGINE on debian 11 and on starting it I am getting the
following errors.
Oct 04 12:14:39 ip-172-31-26-106 systemd[1]: Starting NGCP RTP/media Proxy
Daemon...
Oct 04 12:14:39 ip-172-31-26-106 ngcp-rtpengine-iptables-setup[4655]:
On 30/09/2021 13.34, [ EXT ] Alex Balashov wrote:
On Sep 30, 2021, at 1:32 PM, Richard Fuchs wrote:
On 30/09/2021 13.17, [ EXT ] Alex Balashov wrote:
I’m not sure how the mapping works internally. But whatever the operation is,
is that value stored somewhere or possible to store somewhere so
On 30/09/2021 13.17, [ EXT ] Alex Balashov wrote:
I’m not sure how the mapping works internally. But whatever the operation is,
is that value stored somewhere or possible to store somewhere so as to persist
across restarts in a turn-key way?
AFAICR the node is selected based on a
This should be fixed now.
Cheers
On 20/07/2021 20.23, [ EXT ] Anthony Joseph Messina wrote:
Yep. No problem -- that's why I wanted to ask first. Thanks again. I'll
keep an eye out. -A
On Tuesday, July 20, 2021 6:46:40 PM CDT Richard Fuchs wrote:
I haven't had a chance to look at it yet
I haven't had a chance to look at it yet. There's been a lot of
construction going on in current master (and some more to come) so it's
quite possible that this is fallout from that.
Cheers
On 20/07/2021 19.15, [ EXT ] Anthony Joseph Messina wrote:
Thanks, Henning. I rebuilt with that
On 14/05/2021 11.57, [ EXT ] Barry Flanagan wrote:
In more recent versions of rtpengine you can enable local RTCP
generation and this should give you stats even if the clients don't
send RTCP.
I am running rtpengine 8.5.3.2 - how would I enable this feature or
what version would I need?
Are your RTP clients sending RTCP? MOS calculation depends on RTCP. You
can look in rtpengine's log, at the end of the call the MOS score is
logged together with other stream stats.
In more recent versions of rtpengine you can enable local RTCP
generation and this should give you stats even
On 28/04/2021 10.53, [ EXT ] Володимир Іванець wrote:
Hello!
I'm testing call recording with Rtpengine. It works fine when the
"record-call=on" flag is added to the /rtpengine_offer/ or
/start_recording/ is used in the *request_route*.
But I was wondering if the call recording can be
On 21/01/2021 00.32, mahesh prasad behera wrote:
Hi Team,
We are using sipwise rtpengine on platform centos 7. To get call
related statistics from rtpengine. We tried to use utils
"rtpengine-ng-client" and "rtpengine-ctl". But for us both of them are
not working.
*Rtpengine process
On 19/01/2021 07.11, Daniel-Constantin Mierla wrote:
Hello,
I built rtpengine deb packages for debian just a few days ago and all
went fine. I used the file mr8.5.1.5.tar.gz from the github releases
of rtpengine project.
However, I noticed that some past releases fail to build because of
On 17/12/2020 09.00, Sergio Charrua wrote:
Personally I use NATHelper and RTPEngine modules
RTPEngine, afaik, is the latest and upgraded version of RTPProxy
(please, anyone correct me if that is not the case)
They serve a similar purpose, but they're unrelated products and
rtpengine is not
Use IPv6
Cheers
On 07/12/2020 23.01, David Cunningham wrote:
Hello,
We have a problem with a SIP doorbell device which sends media one way
only, and NAT at the receiving device.
When the doorbell button is pressed it makes a call to a configured
destination. Since the doorbell only
On 04/12/2020 14.04, Andrew Chen wrote:
Sure...I understand ICE has its own setup workflow than SIP but it's
also important that rtpengine uses the rtp path that's negotiated in
the SIP or else it can cause confusion (to those who don't understand
ICE very well like me).
There is no RTP path
On 04/12/2020 13.36, Andrew Chen wrote:
Hmm..that's interesting. You would guess that the rtpengine binary
process shouldn't start connecting ICE candidates once the SIP part is
fully negotiated, which should trigger the rtpengine module on the
Kamailio to tell rtpengine binary.."ok..you can
On 04/12/2020 13.10, Andrew Chen wrote:
So from a SIP point of view, the 200 OK should of sent the final
negotiation of SDP once the client ACK's it right?
The requirement to send an updated offer once ICE has completed with the
final negotiated candidates existed in the original ICE RFC, but
On 04/12/2020 11.39, Andrew Chen wrote:
oh...that's the IPv6 address of the STUN server, not the ipv6 of the
rtpengine instance.
Ok. From rtpengine's point of view, this is one of the client's IP
addresses. The ICE candidate in the SDP is the client telling rtpengine:
This is one of my own
On 04/12/2020 11.31, Andrew Chen wrote:
If that's the case then I don't know why this line doesn't show the
ipv6 address of the client:
Dec 3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq
On 04/12/2020 10.39, Andrew Chen wrote:
So my next question would be:
My 200 OK back to the client should have rtpengine as the only ICE
candidate. Shouldn't it use that one instead?
Yes it should. And it probably does.
Cheers
___
Kamailio
On 04/12/2020 09.40, Andrew Chen wrote:
Hey Richard,
So that's what I thought too until I saw this in rtpengine logs for
one of my test calls:
Dec 3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG:
[ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint:
2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827
On 04/12/2020 09.24, Andrew Chen wrote:
Hey Richard,
So it is true rtpengine is handling rtp between kamailio and receiver
(freeswitch). I'm trying to understand if there is a way to not
forward rtp to any of the ICE candidates in the original INVITE
request from the client side. In other
On 04/12/2020 09.11, Andrew Chen wrote:
So for Yuriy's comment:
I did issue ICE=force parameter but, as you can see my paste, it's
still sending RTP sequence packets to the ICE candidate, which is not
what I want to do.
Richard,
So our current setup is this:
SIP
client -> kamailio ->
On 03/12/2020 13.39, Andrew Chen wrote:
Hi all,
I was wondering if someone can help me understand how the ICE
parameter works in the rtpengine module works.
So basically our client does an ICE candidate lookup and grabs a list
of them and applies it to the INVITE that gets sent to the
Can't say for sure from the log you posted, but my guess is that you're
trying to play media to the B side before having received an answer from
B. Enable debug logging for a better view of what's happening.
Cheers
On 21/10/2020 11.43, Amit Pal wrote:
Hi,
I have using kamailio with
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&quo
On 12/09/2020 05.51, Gholamreza Sabery wrote:
Hi,
I checked rtpengine's documentation, and I wonder what happens if I
use SRTP-DTLS with rtpengine? My connection will be hop by hop
encrypted with rtpengine, or like TURN my connection will remain
end-to-end encrypted??
In most cases,
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
On 02/09/2020 19.30, Patrick Wakano wrote:
Hello list,
Hope you are all well.
Under some load test simulations I've been facing cases where the
command Kamailio sends to RTPEngine times out with such message:
send_rtpp_command(): timeout waiting reply for command "" from RTP proxy
After
On 22/08/2020 09.39, Amit Pal wrote:
Dear Team,
I am using kamailio with rtpengine for media.
When I make call between two sipUA then media flow is manage by
rtpengine offer/answer.
The issue is when hold the call then following some error are coming .
Pls find the full
On 20/08/2020 03.42, Amit Pal wrote:
Dear Team,
I am using kamailio with rtpengine for media.
when call getting hold no media sound play (configure hold.mp3).
From rtpengine gets following error.
rtpengine[624]: ERR: No suitable SDP section for media playback
rtpengine[624]: WARNING:
On 10/07/2020 04.59, Benjamin Flügel | vio:networks wrote:
Hey guys,
I'm trying to configure a Kamailio to work with a browser softphone based on
SIPJS using WebRTC.
So far it works great on Firefox but have a specific problem with chrome, when
I want to make call from the softphone to
, Andrew Chen wrote:
Thanks Richard.
I do have the nathelper module running but not rtpproxy. How does
nathelper cause this issue?
On Fri, Jun 26, 2020 at 8:27 AM Richard Fuchs <mailto:rfu...@sipwise.com>> wrote:
Looks like you're using both rtpengine and some other
SDP-modifyi
Looks like you're using both rtpengine and some other SDP-modifying
module together, such as nathelper or rtpproxy, without calling
msg_apply_changes() in between.
Cheers
On 25/06/2020 16.46, Andrew Chen wrote:
Hi forum,
I'm starting my rtpengine project and I'm facing a strange problem
On 22/01/2020 14.26, Daniel-Constantin Mierla wrote:
Btw, a few questions for further clarifications: if a call has
parallel forking and rtpengine offer is executed for each branch, one
is answered, but the other branch-sessions are not not deleted, what
does rtpengine? It times them out, or
On 22/01/2020 13.15, George Diamantopoulos wrote:
On Wed, 22 Jan 2020 at 18:28, Richard Fuchs <mailto:rfu...@sipwise.com>> wrote:
On 22/01/2020 11.06, Sebastian Damm wrote:
> Hi,
>
> our scenario is the following: We have two clients registered to our
On 17/01/2020 09.01, Daniel-Constantin Mierla wrote:
Hello,
do people here have (implemented) special ways to properly start
rtpengine with kernel forwarding after system reboot?
On our own systems, we have xt_RTPENGINE loaded through modules-load.d,
and the iptables rule added through
On 16/01/2020 12.29, Nuno Miguel Reis wrote:
Hi again.
Thanks for all the help and suggestions. I realized the issue happens
if using kernel forwarding only. If I change rtpengine to start at
userspace without the kernel module enabled everything works fine as
expected.
Do you have any hints
On 15/01/2020 13.39, Nuno Miguel Reis wrote:
Hi guys.
I'm replacing a environment where I was using kamailio + freeswitch by
another where I'm adding rtpengine to the mix.
One of the issues I'm having now is when I have a SIP Client behind NAT:
When I send the INVITE from the SIP Client, the
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
On 29/05/2019 06.57, Carsten Bock wrote:
Hi,
quick question:
If understood it correctly in the past (and it worked for me quite
well this way), I can simply call rtpengine_manage() for subsequent
requests without adding any additional information (e.g. used
interfaces, RTP/SRTP translation
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
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")) {
`replace-origin` and `replace-session-connection` are optional for
normal operation and are preempted by `ICE=force-relay`. If you want c=
and o= (and m=) lines updated, you should use `ICE=force` instead.
Cheers
On 19/04/2019 05.25, David Dean wrote:
Sorry, not sure I understand.
The
Hi,
(X-posted to sr-dev as this is getting into the nitty gritty)
As a short-term workaround for this, I've been playing with the
preloaded library approach to hijack the pthread mutex calls and force
them to provide process-shared mutexes. AFAICT this seems to be working
and only has the
On 01/04/2019 09.14, Istvan Mogyorosi wrote:
Dear all,
This is my first post after reading a lot in this mailing-list.
I'm trying to use Kamailio 5.1 with the dispatcher module and
rtpengine acting as SIP + RTP proxy.
I have 6 asterisk servers in a private subnet that should talk with
the
On 01/04/2019 11.22, Fred Posner wrote:
On 4/1/19 11:18 AM, Steve Davies wrote:
Never quite got this. UDP packets can be up to 64k, right? And
fragmentation is a standard IP feature if a packet is bigger than MTU
size.
Steve
of course, you're hoping that the fragmented packets arrive in
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
On 25/02/2019 13.40, Daniel-Constantin Mierla wrote:
I will look into this direction as well, there was something reported
also for t_should_relay_response() over the time.
You were running 5.2.1?
This one was on 5.1.7.
Cheers
___
Kamailio (SER)
On 25/02/2019 12.34, Daniel-Constantin Mierla wrote:
Hello,
that's strange, but a while ago someone else reported an issue with
same backtrace.
So the crash happens at the last line in the next snippet from
reply_received() function in the tm module:
uac=>uac[branch];
On 20/02/2019 06.53, Prabhat Kumar wrote:
I am getiing this error in syslog
Feb 20 11:48:58 ip-10-0-0-X systemd[1]: Started Cleanup of Temporary
Directories.
Feb 20 11:49:39 ip-10-0-0-X systemd[1]: Stopped NGCP RTP/media Proxy Daemon.
Feb 20 11:49:39 ip-10-0-0-X systemd[1]: Starting NGCP
On 16/02/2019 07.38, vinod mn wrote:
Hi I am not able to do SRTP configuration properly
this the kamailio configuration
*route[NATMANAGE] {
*
*rtpengine_manage("replace-origin replace-session-connection RTP/SAVP");
}*
I can see in SDP RTP/SAVP but still the packets are RTP.
clients used
On 06/02/2019 01.55, Laurent Schweizer wrote:
Hello,
I’m using the RTPengine with Kamailio and I have a question for a
specific case.
I have some customer that are changing the source port of the RTP
stream during the call ( no re-invite) I think it’s more a NAT issue
that a user agent
On 02/01/2019 09.32, Yuriy Gorlichenko wrote:
Thx for the reply
Yes
Internal hash table diffenentelly stores info
But even it case of putting timeout to 0 it still grows in synthetic
tests. So looks like it will grows alsways because of deletes entries
but creates new and so on and so on...
On 02/01/2019 07.45, Yuriy Gorlichenko wrote:
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
On 06/12/2018 07.17, Mark Hall wrote:
Hi,
I am using rtpengine to proxy audio to/from media servers.
The outbound calls are being rejected by a couple of my carriers
(although accepted by others), and the closest that I have been able
to find for the reason is the inclusion of
This is usually a symptom of doing a double SDP rewrite, e.g. by making
a call to rtpengine twice in the same script iteration, or making a call
to rtpengine and then doing some other SDP rewrite operations either
before or after that.
Cheers
On 2018-10-23 21:54, Wilkins, Steve wrote:
On 2018-09-06 06:13, Carsten Bock wrote:
Hi Richard,
this is awesome! Thanks!
One question:
Currently, the XML-RPC URL has to be set in the config, right? I
frequently have the following setup:
- 2x Kamailio
- 2x RTPEngine
(and each Kamailio is connected to both RTPEngines, for redundancy and
,
Daniel
On 05.09.18 08:52, Richard Fuchs wrote:
Yup that's exactly right.
It would be fairly simple to implement an additional XMLRPC format if
there's a particular one that's more friendly towards Kamailio.
Cheers
On 2018-09-05 02:42, Daniel-Constantin Mierla wrote:
Looking quickly
handle it using xmlops, xhttp and jsonrpcs modules.
Cheers,
Daniel
On 05.09.18 08:24, Richard Fuchs wrote:
It does an XMLRPC callback. Currently there's two formats for it, one
is a sems sbc teardown request (using the from-tag), the other is a
generic "teardown" command using t
It does an XMLRPC callback. Currently there's two formats for it, one is
a sems sbc teardown request (using the from-tag), the other is a generic
"teardown" command using the call ID.
Cheers
On 2018-09-04 07:52, Daniel-Constantin Mierla wrote:
Hello,
what do you get from rtpengine on rtp
It may be more helpful to post some logs from rtpengine. You should
never see "*Call-ID not found*" from an offer.
Cheers
On 2018-08-22 08:49, Wilkins, Steve wrote:
Here is my start up =>
rtpengine --interface 111.121.22.11\!27.22.132.10 --listen-ng
127.0.0.1:12221 --dtls-passive -f -m
On 2018-08-16 05:23, Mojtaba wrote:
Hello,
Before i used RTPEngine (6.4.0.0+0~mr6.4.0.0) without any problem,
Nowday i install RTPEngine from git. I notice that it's varsion is
changed to 6.5.0.0+0~mr6.5.0.0. When i want install
ngcp-rtpengine-recording-daemon, I have this issue:
Failed to start
On 2018-08-14 04:45, Nicolas Breuer wrote:
I don't like the msg_apply_changes but can you explain the different behavior
between rtpproxy & rtpengine?
The difference is due to the fact that rtpengine rewrites and replaces
the entire SDP body, while rtpproxy only manipulates bits and pieces
On 2018-08-08 13:05, Alex Balashov wrote:
On Wed, Aug 08, 2018 at 09:38:02AM -0400, Richard Fuchs wrote:
On 2018-08-08 09:25, Alex Balashov wrote:
Richard,
rtpproxy classic edition had an SDP attribute that could be inserted to
prevent rtpproxy from operating on the SDP if another rtpproxy
On 2018-08-08 09:25, Alex Balashov wrote:
Richard,
rtpproxy classic edition had an SDP attribute that could be inserted to
prevent rtpproxy from operating on the SDP if another rtpproxy had
already been engaged upstream. Does RTPEngine have a similar feature?
Yes, if you include the
On 2018-08-06 21:17, Alex Balashov wrote:
From the 5.1 TB transferred on the 'lo' interface (RTPEngine running on
same host as Kamailio):
lo: flags=73 mtu 65536
inet 127.0.0.1 netmask 255.0.0.0
inet6 ::1 prefixlen 128 scopeid 0x10
loop txqueuelen 1 (Local
On 2018-08-06 06:58, Sebastian Damm wrote:
Hi,
we run multiple rtpengine servers to share the load. Whenever we need
to take an rtpengine server offline, we used to just block the control
port via iptables, then no new calls ended up on this instance of
rtpengine. This worked pretty well in
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