ch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
set_contact_alias();
}
}
#!endif
return;
}
- Original Message -----
From: Daniel-Constantin Mierla
To: Marko Tirs ; Kamailio (SER) - Users Mailing List
Sent: Tuesday, March 14, 2017 5:23 PM
ch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
set_contact_alias();
}
}
#!endif
return;
}
- Original Message -----
From: Daniel-Constantin Mierla
To: Marko Tirs ; Kamailio (SER) - Users Mailing List
Sent: Tuesday, March 14, 2017 5:23 PM
Hello,
I think you can call the start recording function just after
rtpproxy_manage().
If you don't use the start recording function, is the audio going both
ways without problems?
Cheers,
Daniel
On 12/03/2017 15:04, Marko Tirs wrote:
> Hello experts,
>
> I'm newbie in Kamailio and have a prob
Hello,
you need to run rtpproxy in bridge mode. See the alg example from the
rtpproxy module -- online at:
-
https://github.com/kamailio/kamailio/blob/master/src/modules/rtpproxy/examples/alg.cfg
Cheers,
Daniel
On 05/01/2017 01:09, Rodrigo Moreira wrote:
> Hello,
>
> I am having problems wit
Yep, logs helps. Actually, was mix of errors in PBX and Kamailio config.
So yea, rtpproxy is working as expected, can’t say same for my config files :)
Regards, Igor
On 22 дек. 2016 г., 19:24 +0200, Alex Balashov ,
wrote:
> That just sounds like the rtpproxy is not being engaged, i.e. that the
Oh, I see. Yes, that could be.
-- Alex
> On Dec 22, 2016, at 12:49 PM, Daniel Tryba wrote:
>
>> On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote:
>> That just sounds like the rtpproxy is not being engaged, i.e. that the
>> rtpproxy_manage() call is failing. When that happens, the
On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote:
> That just sounds like the rtpproxy is not being engaged, i.e. that the
> rtpproxy_manage() call is failing. When that happens, the SDP from .2
> will be passed through unaltered.
>
> The Kamailio log should give you some idea of why
That just sounds like the rtpproxy is not being engaged, i.e. that the
rtpproxy_manage() call is failing. When that happens, the SDP from .2 will be
passed through unaltered.
The Kamailio log should give you some idea of why the rtpproxy invocation has
failed.
RTPProxy is certainly not limited
Sorry, I got media address 111.222.3.2 in SDP, but not 111.222.3.3
Regards, Igor
On 22 дек. 2016 г., 19:19 +0200, Igor Olhovskiy ,
wrote:
> Hi!
> Issue I can’t figure out. Or all working ok and that’s just me who not
> understands.
>
> I have situation
> softPhone (111.222.3.2) -> Kamailio w.
Hi Shantanu,
You need to modify kamailio config to achive your goal. Look into NAT
routing (i.e. NATDETECT and NATMANAGE).
2016-12-03 8:23 GMT+02:00 shantanu saha :
> Hello,
> I am trying to configure rtpproxy with Kamailio. kamailio.cfg is already
> configured (default) for rtpproxy. I just add
Yes. I finish the session at the end of each call. Also I am using
rtpproxy_manage("r"). My configuration regarding RTPProxy is default config
which comes with Kamailio sample config file.
On Tue, Nov 15, 2016 at 11:44 AM, Dragos Oancea
wrote:
> Hi
>
> Increase fs.file-max in your /etc/sysctl.co
Hi
Increase fs.file-max in your /etc/sysctl.conf .
eg: fs.file-max = 5
And then do sysctl -p
Decrease SILENT_TIMEOUT in your rtpengine.conf (eg:SILENT_TIMEOUT=120) -
it's default 1 hour and if some calls don't have media then rtpengine
will just keep the UDP ports in use until this timeout exp
Are you ending the rtp proxy sessions when the call are ended? What
rtpproxy functions are you using in the configuration file?
Cheers,
Daniel
On 14/11/16 18:18, Gholamreza Sabery wrote:
> No not the first time. But over time. I rebooted my system and error
> is gone! It seems that it happens ov
No not the first time. But over time. I rebooted my system and error is
gone! It seems that it happens over time.
On Mon, Nov 14, 2016 at 6:12 PM, Daniel-Constantin Mierla wrote:
> Are you getting the error first time when you reach first 1900 sessions?
> Or after a while, after some previous se
Are you getting the error first time when you reach first 1900 sessions?
Or after a while, after some previous sessions are ended?
Cheers,
Daniel
On 14/11/16 11:19, Gholamreza Sabery wrote:
> I already set these parameters:
>
> rtpproxy -m 5000 -M 65000
>
> As well limits for number of open files
I already set these parameters:
rtpproxy -m 5000 -M 65000
As well limits for number of open files are set to 100 (ulimit -n).
When I increased log level of RTPProxy I saw:
ERR:create_twinlistener:GENERAL: can't create IPv4 socket: Too many open
files in system
On Mon, Nov 14, 2016 at 1:46 P
Hello,
first thing to look at is the port range. There are some parameter that
you can provide to rtpproxy in command line in order to increase the
range of port it can use -- see 'rtpproxy -h' or 'man rtpproxy'.
Cheers,
Daniel
On 14/11/16 11:14, Gholamreza Sabery wrote:
> Dear Daniel:
>
> I us
Dear Daniel:
I used a single RTPProxy instance. RTPProxy version = 20040107. And yes
there was traffic for all calls but traffic is one-way. One leg sends the
call and the other just receives it.
On Mon, Nov 14, 2016 at 1:40 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> have you used a singl
Hello,
have you used a single rtpproxy instance? Was there RTP traffic for all
1900 calls? Is this with rtpproxy 1.2 or 2.0?
Cheers,
Daniel
On 14/11/16 10:44, Gholamreza Sabery wrote:
> I managed to create about 1900 concurrent calls using a single
> Kamailio and RTPProxy server. But after this
Hello,
thanks for all those details, very useful ...
To be clear -- the issue of using high cpu on idle (no active calls) was
with rtpproxy v1.2 on a centos (iirc, v6), not with rtpproxy 2.0. On
debian, same version of rtpproxy was not exposing this. I was just
curios to see if anyone else saw it
My personal opinion on this is that it should be very low-priority. It's
one of those problems that takes 99% effort for 1% marginal results, and
even then, rather imperfect ones.
For almost any service provider, having a media relay while calls are on
it is not the worst possible problem—cert
Just a little comment on the numbers that I've thrown out earlier today.
Those are probably somewhat pessimistic, with some creative tuneup you can
probably go much higher. But we also constrained by some other
considerations (i.e. running fully redundant network connection with FEC,
full firewall
Arsen, there is no readily-made solution with rtpproxy unfortunately for
that. Some time around 1.0 times circa 2007-2009, somebody submitted a very
rough patch to implement master/hot-standby scenario, but the patch was not
production-ready back then and the contributor was not available to refine
Daniel, thank you for your interest. Yes, there were many architectural
changes between 1.x and 2.0. The most noticeable is that we've decoupled
I/O from the control channel handling and also split I/O into two threads,
one for poll/receive and the second one for the sending. We've also
refactored
Hi guys,
In addition to this interesting and useful thread, what is the best way to
implement media session recovery, for example in Active/Passive HA scenario?
I know that it is possible with rtpengine (redis db), is it possible with
rtpproxy?
Thanks,
Arsen.
On Wed, Oct 19, 2016 at 11:19 AM, Da
Hello Maxim,
given the discussion here, I would like to get some updates for myself
regarding 2.0 in terms of capacity and other stuff.
I was using rtpproxy 1.x with kamailio doing load balancing across many
instances of rtpproxy. I was using 1000 streams as estimation for one
instance and I see
And yes, I was remiss in failing to mention that an effective solution
to scaling out rtpproxy is to bind multiple instances with different
core affinities.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristes
Alex, no problem. Nobody knows everything. :)
-Max
On Wed, Oct 19, 2016 at 12:35 AM, Alex Balashov
wrote:
> Hi Maxim,
>
> Duly noted! I certainly did not intend to mislead anyone or to be
> disingenuous; I gave information that was, to the best of my knowledge,
> true. I appreciate your followu
Hi Maxim,
Duly noted! I certainly did not intend to mislead anyone or to be disingenuous;
I gave information that was, to the best of my knowledge, true. I appreciate
your followup and clarification, which certainly is useful for my own knowledge
as well!
My sincere apologies...
-- Alex
O
Alex, with all due respect, things you said about rtpproxy capacity is
somewhat outdated and misleading. We have some nodes in the field, that
handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy
instances, 1,000 sessions each. 2-3 year old CPUs, 12 cores in total.
We also have a
On 10/17/2016 02:29 PM, Rodrigo Moreira wrote:
What is difference between modules rtpproxy and rtpengine?
rtpproxy is a userspace process which, historically, has a relatively
limited call throughput capacity (maybe a few hundred calls), though
this might be addressed to some degree in rtppr
Hi again
Yes, i feel very dumb because i actually faced this problem in my testing
with SIPp and with Asterisk before and it was all about open file
description. I probably didnt think about it because I couldnt find
anything in the logs about it which it is quite weird, the only thing that
logs i
Yes, that's probably it. There should be also some error in the log that
rtpproxy emits, so you might want to check that. I see people run into this
from time to time, perhaps we need to check and put out a big warning in
red if the OS limit appers to be too low?
-Max
On Sun, Jul 17, 2016 at 3:18
On 07/17/2016 05:49 PM, TEG AMJG wrote:
Dear list
I am quite new to Kamailio and i have been able to solve some NAT
Traversal issues with symmetric SIP+RTP putting kamailio+rtpproxy
behind NAT, i am also load balancing some asterisk boxes for
transcoding and some other services like voicemail
We don't support SRTP de/re-encryption at this point, neither in master nor
in 2.0. The work to add it is underway, but we are not quite there yet.
Pass-through mode should be working fine though, we've tested it recently
specifically.
On Jun 7, 2016 12:27 PM, "Albert Petit" wrote:
> Hi ,
>
> Sor
Hi ,
Sorry for previous question. Finally it seems i got confused because
traffic my UA was sending was not properly encrypted .Then when doing
server tcpdump wireshark was seeing it still as RTCP (and not SRTCP) when
jumping on rtpproxy :-) When fixed problem in user agent all traffic is
SRTCP an
Hello,
are you sure rtpproxy 2.0 does encryption/decryption of the RTP/SRTP? I
haven't noticed that the v2.0 has support for such feature.
What are the parameters you are using to control rtpproxy from kamailio.cfg?
Cheers,
Daniel
On 01/06/16 12:18, Albert Petit wrote:
> Hi list
>
> I have ins
Hello,
I am not familiar with the insights of rtpproxy source code, so I don't
know if there is a limitation for duration and for how long it is.
Cheers,
Daniel
On 27/05/16 10:42, gmele wrote:
>
> Hello,
>
>
>
> Thx for the explanation, so it means that as soon as the callee
> connects to the
Hello,
Thx for the explanation, so it means that as soon as the callee connects to the
RTP Proxy, the rtp proxy will use the callee ip address and port to forward the
rtp stream and ignore the initial learned ip/port? Is there a duration
limitation in this learning mode? Meaning that if the cal
Hello,
initially the rtpproxy is in so called learning mode, waiting for the
first rtp packet to come from each side of the call. Before receiving
first rtp packet it relies on source ip of signaling.
If the SDP has the device IP (you can eventually set that in the proxy),
then you can use 'r' fl
Hello,
if rtpproxy is listening on udp:127.0.0.1:7722, then you may have some
limits in place regarding sockets/traffic -- if you have selinux
enabled, try without it. Also, you can try by lowering the value for
children parameter.
Regarding the SQL errors, it seems you run the config from Siremi
Hi Rehan,
No matter which mode you are running rtpproxy in that IP will always be the
IP of the machine it is running on.
That means that SDP will take that IP once routed to locally subnet A2B
servers.
As far as the A2B detecting SIP user as online or offline based on DB, I
am not too sure about
I solved this problem by changing the 127.0.0.1 in rtpengine startup
config into 10.109.247.90 which is the eth0 address of the rtpengine
machine.
Now Kamailio successfully connect to the rtpengine daemon.
Nice you solved it. Didn't think of it :D
___
I’ve already set modparam("rtpengine", "force_send_interface",
“10.109.247.80”) in my kamailio.cfg from the very beginning.
Can you try force ping from kamailio machine (.80) to rtpengine machine
(.90), on 10.109.247.80 interface? (it should work)
There is nothing prompt out on command “kamc
I solved this problem by changing the 127.0.0.1 in rtpengine startup config
into 10.109.247.90 which is the eth0 address of the rtpengine machine.
Now Kamailio successfully connect to the rtpengine daemon.
> 在 2015年12月22日，17:38，Zodiac 写道：
>
> Hi, I am very glad that you can answer me for tha
Hi, I am very glad that you can answer me for that.
I’ve already set modparam("rtpengine", "force_send_interface", “10.109.247.80”)
in my kamailio.cfg from the very beginning.
The port 7723 on 10.109.247.90 which rtpengine daemon runs is not blocked by
firewall.
There is nothing prompt out on
And, I also tried to set Kamailio and rtpengine daemon in a same
machine,and use modparam(“rtpengine”, “rtpengine_sock”,
“udp:localhost:7723”). And Kamailio can work functionally under this
situation. rtpengine daemon can receive ping message from Kamailio and
rtpengine daemon can work as suspe
Hi Zodiac,
1. Can you see your configured rtp node on: "kamctl fifo nh_show_rtpp all" ?
2. Can you ngrep the commands being sent by kamailio to rtpengine?(on
both kamailio and rtpengine machine)
3. Do you have some firewalling rules that may block that 7723 port?
Stefan
On 21.12.2015 17:45, Z
Yes, you can certainly run rtpengine on a different host than the
Kamailio host.
Is it possible there are firewall rules on the remote rtpengine host
that are blocking UDP connections from your Kamailio box to port 7723?
That would be my first thought.
-- Alex
--
Alex Balashov | Principal
Hi Javohir,
don't worry, your english is fine.
Can you send us the parameters you have set for rtpproxy in your kamailio cfg?
You should find something like this:
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:2")
Make sure it matches the configuration of the RTPProxy.
Thanks,
Carsten
Daniel,
Clear, i will simply add the field until we can upgrade. Thanks for the
clarification.
Jan
Daniel-Constantin Mierla schreef op 2015-11-19 13:38:
Hello,
it was a different value for column name in the code than in the
default
rtpproxy table. The definition of the table used setid a
Hello,
it was a different value for column name in the code than in the default
rtpproxy table. The definition of the table used setid and the code
set_name. I pushed a fix (in master, 4,3 and 4,2 branches), so the code
is coherent with the database table.
If you don't want to upgrade to latest v
Hello,
the code shows that is added for U command, if to-tag exists:
https://github.com/kamailio/kamailio/blob/master/modules/rtpproxy/rtpproxy.c#L2692
Can you edit the rtpproxy.c and add a log message (e.g., using
LM_ERR("..."); ) there and is it is printed by kamailio?
Cheers,
Daniel
On 17/1
Looks like I forgot to call rtpproxy_manage() for BYE and CANCEL requests.
netstat -unlp showed a lot of port being used by rtpproxy even after the load
test finished.
The ports weren't being freed and that's probably the reason it gave the error.
It was running out of ports to use.
From: sr-us
Have tested it on virtual and physical. Works well, no need to patch for VM or
advertised address. Had no complaints from users with 1.2 and none since 2.0
Installed from git for testing.
-- Fred
> On May 22, 2015, at 4:37 AM, Klaus Darilion
> wrote:
>
> Hi!
>
> I just found out that ther
On 22.05.2015 10:37, Klaus Darilion wrote:
I just found out that there is a new rtpproxy release:
http://www.rtpproxy.org/post/v2release/
Has anybody tested it and want to share some experiences? Or have people
turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do
not need new
On 18/05/15 03:53 AM, Sebastian Damm wrote:
Hi Alex,
On Thu, May 14, 2015 at 5:47 AM, Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:
According to the rtpengine module documentation for
rtpproxy_manage(), that's exactly what rtpproxy_manage() does:
http://kamailio.org/doc
Hi Alex,
On Thu, May 14, 2015 at 5:47 AM, Alex Balashov
wrote:
> According to the rtpengine module documentation for rtpproxy_manage(),
> that's exactly what rtpproxy_manage() does:
>
>
>
> http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.rtpengine_manage
>
> i.e.
>
> -
Hello Sebastian,
On 05/13/2015 10:12 AM, Sebastian Damm wrote:
Is there a way to have rtpproxy_manage() handle those calls
automatically? I was thinking of setting a flag in the request which
is used by the rtpproxy_manage() function later.
According to the rtpengine module documentation for
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" wrote:
> Yes
>
> On Tuesday, 17 March 2015, Zheng
th the new
version. I will let you know if I reproduce the issue with the new version.
Regards,
Igor.
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Maxim Sobolev
Envoyé : lundi 9 mars 2015 19:09
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPP
Do you mean ROHC ?
2015-03-14 12:39 GMT+08:00 Maxim Sobolev :
> Do you have any particular RFC in mind?
> On Mar 12, 2015 10:28 AM, "John Mathew"
> wrote:
>
>> Hi,
>>
>> Maxim,
>> Is there any plans for rtp header compression in future. I can't see
>> anything in the change log for 2.0.0
>>
>>
>
Do you have any particular RFC in mind?
On Mar 12, 2015 10:28 AM, "John Mathew" wrote:
> Hi,
>
> Maxim,
> Is there any plans for rtp header compression in future. I can't see
> anything in the change log for 2.0.0
>
> On Tuesday, 10 March 2015, Maxim Sobolev wrote:
>
>> Hi All,
>>
>> I'm happy t
e* Maxim Sobolev
> *Envoyé :* samedi 7 mars 2015 09:14
>
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
> Ah, ok, I see now. I did not realize you guys are using resizer. Which
> version of the software are you actually using? I.e. i
Hello,
use rtpproxy_manage() instead of force_rtp_proxy() -- that function was
replaced in newer versions.
Cheers,
Daniel
On 06/03/15 11:53, Tom Braarup Cuykens wrote:
> kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
>
> I found this guide for an older version of Kamailio, but t
lev
Envoyé : samedi 7 mars 2015 09:14
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy issue?
Ah, ok, I see now. I did not realize you guys are using resizer. Which version
of the software are you actually using? I.e. is it latest rel_2_0 / master, or
some legacy 1.x c
ards,
>
>
>
> Igor.
>
>
>
> *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *De la
> part de* Maxim Sobolev
> *Envoyé :* vendredi 6 mars 2015 07:44
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] RTPProxy issue?
>
>
>
&
mail.com
<mailto:igor.potjevle...@gmail.com> ]
Envoyé : jeudi 5 mars 2015 11:34
À : mico...@gmail.com <mailto:mico...@gmail.com> ; 'Kamailio (SER) - Users
Mailing List'
Objet : RE: [SR-Users] RTPProxy issue?
Hello,
Thank you.
Just to let you know, the RTPProxy
..@gmail.com]
> *Envoyé :* jeudi 5 mars 2015 11:34
> *À :* mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
> *Objet :* RE: [SR-Users] RTPProxy issue?
>
>
>
> Hello,
>
>
>
> Thank you.
>
>
>
> Just to let you know, the RTPProxy is running in
"True" is coming from the UA.
The RTPProxy stops forward since this packet.
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
Envoyé : jeudi 5 mars 2015 11:34
À : mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Us
: [SR-Users] RTPProxy issue?
Hello,
maybe Maxim (cc-ed) will be able to provide more insights.
Cheers,
DAniel
On 04/03/15 16:59, Igor Potjevlesch wrote:
Hello,
I discovered an issue related to the handling of "timestamp" and/or "Marker
bit" with rtpproxy (I use
Hello,
maybe Maxim (cc-ed) will be able to provide more insights.
Cheers,
DAniel
On 04/03/15 16:59, Igor Potjevlesch wrote:
>
> Hello,
>
>
>
> I discovered an issue related to the handling of "timestamp" and/or
> "Marker bit" with rtpproxy (I use the latest Extension 20081224).
>
>
>
> The c
On Feb 16, 2015, at 7:27 PM, Ovidiu Sas wrote:
> You could simply let the RTP traffic to flow directly between FS and
> endpoints (no need for rtpproxy).
> All you need to do is:
> - forward the appropriate RTP ports to FS;
> - fix the private IP in SDP by replacing it with the public IP for
> t
Could you show the revelant codes in rtpproxy and kamailio.cfg? I am unable
to get the audio pass through from extranet to intranet as private IP
address is used after rtpproxy.
I use Kamailio 4.2 and rtpproxy in Debian wheezy. Both are installed from
repository.
On Tue, Feb 17, 2015 at 8:27 AM,
You could simply let the RTP traffic to flow directly between FS and
endpoints (no need for rtpproxy).
All you need to do is:
- forward the appropriate RTP ports to FS;
- fix the private IP in SDP by replacing it with the public IP for
the inbound rtp streams (to FS).
-ovidiu
On Mon, Feb 16, 20
On 16/02/15 01:12 PM, Virmantas Variakojis wrote:
> Could you provide us a little example? For examlple i have kamailio with
> three interfaces: two interfaces (vlan's look at two different
> providers) and third interface looks at sip clients.
You would define two interfaces with different names,
Could you provide us a little example? For examlple i have kamailio with
three interfaces: two interfaces (vlan's look at two different providers)
and third interface looks at sip clients.
Thank's in advance!
2015 vas. 16 20:04 "Richard Fuchs" rašė:
> On 16/02/15 01:00 PM, Virmantas Variakojis wr
On 16/02/15 01:00 PM, Virmantas Variakojis wrote:
> Hi,
>
> There pathch with -A can be found or it is allready implemented into
> specific rtpengine version?
Latest master from git. The command line syntax is a bit different from
rtpproxy, but the basic idea is the same.
Cheers
___
Hi,
There pathch with -A can be found or it is allready implemented into
specific rtpengine version?
2015 vas. 16 19:50 "Richard Fuchs" rašė:
> On 16/02/15 12:39 PM, Muhammad Shahzad wrote:
> > I haven't done something like that myself but i think if you use
> > RTPEngine with "media-address" se
On 16/02/15 12:39 PM, Muhammad Shahzad wrote:
> I haven't done something like that myself but i think if you use
> RTPEngine with "media-address" set correctly in offer and answer
> functions, you can easily achieve this. Simply check if request/reply is
> coming from FS or the end-user and adjust
BTW, if nothing works, you can always use "network:msg" event route to find
/ replace any part of the SIP request and response, including media IP in
SDP. ;-)
http://kamailio.org/docs/modules/4.2.x/modules/corex.html#async.evr.network_io
Thank you.
On Mon, Feb 16, 2015 at 6:39 PM, Muhammad Sha
I haven't done something like that myself but i think if you use RTPEngine
with "media-address" set correctly in offer and answer functions, you can
easily achieve this. Simply check if request/reply is coming from FS or the
end-user and adjust the media appropriately without even invoking
auto-bri
Hello,
rtpproxy doing bridging requires two network interfaces to work with.
You can try one of the following:
- let freeswitch advertise the public ip for media and skip rtpproxy
completely
- use the second parameter of rtpproxy_manage() to set the advertised ip
address for media and don't confi
On 09/29/14 14:29, Marino Mileti wrote:
> Wow! Do you have an example of how to do that? How I have to modify my
> kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in
> this way?
You don't have to do anything, tags are already included in all the
messages.
cheers
__
On 09/29/14 14:08, Marino Mileti wrote:
> But with from-tag and To-tag it's possible to instruct rtpengine to generate
> new couple of ports for each branch of a call? In the source code of
> rtpengine it seems that it check only the callid parameter
Yes it will. The call-id is only a vague umbrel
On 09/26/14 16:57, Marino Mileti wrote:
> Hello,
>
>> On Friday 26 September 2014 16:44:44 Marino Mileti wrote:
>>> Hi guys,
>>> I've seen that setting the parameter extra_id_pv, every branch should
>>> be a different callid..
>>> How can i set this parameter? I've tried with :
>>> modparam("rtpp
On Friday 26 September 2014 16:44:44 Marino Mileti wrote:
> Hi guys,
> I've seen that setting the parameter extra_id_pv, every branch should be hae
different callid..
> How can i set this parameter? I've tried with :
> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)")
>
> but in the INVITE me
On 23/09/14 04:35 AM, marino.mil...@alice.it wrote:
I've a problem with rtpproxy during a parallel ring scenario.
I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I
try to call them in parallel mode (ringall) rtpproxy module sends in to
the INVITE the same RTP ports.
Is it po
.@lists.sip-router.org] De la part de Carsten Bock
> Envoyé : jeudi 22 mai 2014 20:08
> À : Kamailio (SER) - Users Mailing List
> Objet : Re: [SR-Users] RTPProxy timeout
>
> Hi Igor,
>
> it worked pretty well at the time i wrote the patch.
> I've sent the patch to the d
't succeed in patching your rtpproxy and
> use the timeout functionality?
>
> Regards,
>
> Igor.
>
> -Message d'origine-
> De : sr-users-boun...@lists.sip-router.org
> [mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten
> Bock Envoyé
roxy and
> use the timeout functionality?
>
> Regards,
>
> Igor.
>
> -Message d'origine-
> De : sr-users-boun...@lists.sip-router.org
> [mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten Bock
> Envoyé : jeudi 22 mai 2014 18:25
> À : Kamai
ctionality?
Regards,
Igor.
-Message d'origine-
De : sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] De la part de Carsten Bock
Envoyé : jeudi 22 mai 2014 18:25
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] RTPProxy timeout
Hi Igor,
Hi Igor,
in order to use this feature, you need to apply a patch to RTPProxy.
The patch can be found in the "test" subdirectory in modules/rtpproxy/.
I tried to get the patch into the rtpproxy, but i never got an answer
from the developers of RTPProxy.
Kind regards,
Carsten
2014-05-22 11:21 GMT
On Wednesday 23 April 2014 13:41:42 Alex Balashov wrote:
> Just of curiosity, how exactly does the "havoc" manifest itself? What
> are the symptoms of this being a problem?
Havoc might be a bit strong (though "great confusion and disorder" fits well).
Bad sound quality (if any sound) for a longe
On 04/23/2014 07:36 AM, Daniel Tryba wrote:
I'm having some troubles with a provider sending RTP before a 183
Session Progress or 200 OK (I see up to 1s of rtp prematurely). The
machine is running rtpproxy and apparently rtpproxy buffers these rtp
packets and flushed them in one burst when the 1
Dear Daniel,
My whole set-up remains same in both the cases, only change is running
RTPproxy with kamailio server and without running RTPproxy server.
Now When i run RTPproxy with kamailio server, RTP packets has to reach up to
RTPproxy server to get relay to other end client right ?
But how this
Hi,
When you don't use RTPproxy, RTP traffic is sent end-to-end.
What is the RTP path in case of client-to-client RTP ?
Is it the same ?
Daniel
On 02/25/2014 06:34 PM, Ravi wrote:
> Dear Daniel,
>
> Thank you for the response,
>
>> Do you have any router, firewall something between clients
Dear Daniel,
Thank you for the response,
>Do you have any router, firewall something between clients and mediaproxy ?
Ya I have Routers And switch in between clients and Rtpproxy, My set-up is
as follows:
SIP Clients<-->WAP+WiMax CPE <--> WiMax Base station <>
RTPproxy+Kamailio se
Hi Ravi,
if you media/rtp proxy is receiving such packet loss, it means that
something behind him is cutting the traffic off, somehow.
So you should investigate there, in their configuration.
If your analysis is right, it seems your loosing around 80% of your packets.
Do you have any router, firew
Dear Daniel,
Thank you again,
Ya i am investigating on this issue, by the way can you give any comments on
my questions in the previous mail? I just wanna clarify those things to
rectify this packet loss issue. i googled about those questions but still
ended with the same confusion status and did
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