On 08/06/12 06:08, Vijay Thakur wrote:
Hi all,
I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking
translation and bridging is fully supported. There's no
internal/external IP parameters because you can bridge between any interface,
not only two.
I'm CC-ing the mail to the developer, Richard Fuchs. I'm not sure he reads
this mailing list.
I do, but didn't want to interrupt. ;)
We use
On 09/17/12 11:03, Carsten Bock wrote:
Hi Richard,
i've noticed the use of the i and e flags for your mediaproxy implementation.
Right now i'm testing a minor patch for the RTPProxy module to allow
automatic selection of ie or ei just for this use case based on
the media-ip type (e.g.
Hi,
While I can't really answer your question, the logic in mediaproxy-ng is
that if the to-tag is given in the Delete message, it has to match the
to-tag that was previously given in the Lookup message alongside with
the from-tag. If no to-tag is given in the delete message, then only the
On 10/19/12 11:16, Juha Heinanen wrote:
i have not see in any document description about how long rtp proxy
keeps the call state after it has received US command, but no matching
LS command. is there a timer that clears those hanging calls once in a
while and sip proxy config writer does not
On 10/24/12 14:28, Juha Heinanen wrote:
this is not the real fix, but helps until someone figures out why dns
query on something that is not a name but wrongly formatted ipv6 address
is done in the first place.
What do you mean with wrongly formatted?
cheers
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On 10/24/12 14:41, Juha Heinanen wrote:
Richard Fuchs writes:
this is not the real fix, but helps until someone figures out why dns
query on something that is not a name but wrongly formatted ipv6 address
is done in the first place.
What do you mean with wrongly formatted?
the ipv6
On 10/24/12 14:51, Juha Heinanen wrote:
Richard Fuchs writes:
IPv6 addresses are supposed to be bracketed when used within an URI.
Otherwise a parser wouldn't be able to tell if an optional port was
given or not. Compare http://2620:0:2d0:200::8/ vs
http://[2620:0:2d0:200::8]/ vs http
Hi,
On 11/12/12 13:57, Carsten Bock wrote:
- is there a changelog availabe somewhere?
The changelog is installed through the debian packages in
/usr/share/doc/$PACKAGE/changelog.gz, while the source .tgz contains it
in the ../debian/changelog file.
- i haven't looked at the sources, but can
Hi,
On 04/01/13 10:03, Aft nix wrote:
I stumbled upon this git://github.com/sipwise/mediaproxy-ng.git which
looked very neat to me. Its said that it can be used with kamailio. It
seems like its backed sipwise inc.
But no documentation is given there. Anyone know of a
On 04/02/13 09:15, aft wrote:
So i was asking how to install mediaproxy-ng itself?
If you're on a Debian system, you can simply issue dpkg-buildpackage and
then install the packages it produces. Otherwise you can compile the
sources yourself. A simple make in each one of the 3 subdirectories
On 04/02/13 10:02, aft wrote:
Daemon installation failed with the following :
call.c:15:27: fatal error: xmlrpc_client.h: No such file or directory
Check out the list of dependencies in the debian/control file. One of
them is libxmlrpc-c3 (from http://xmlrpc-c.sourceforge.net/).
What is
On 04/02/13 10:54, aft wrote:
I was actually asking How it works? I mean when there is kernel based
forwarding is enabled, what does the daemon do compared to when the kernel
based forwarding is not enabled?
If i want do some modifications on rtp packets and intend to use kernel
based
On 04/02/13 17:39, aft wrote:
So the bottom line is i have to include the code in both places.
Another thing is i'm assuming you know much about the development of this
media
relay. So i'm asking, is there any plan for including repacketization
feature?
Its very crucial for our
Hi,
On 04/04/13 14:58, Daniel-Constantin Mierla wrote:
quite interesting, I didn't know it has two operations modes: user space
forwarding and kernel forwarding.
Is there any plan in supporting more one mode (or dropping the other) in
the future?
Not per se, kernel mode forwarding (at
On 04/05/13 03:53, Daniel-Constantin Mierla wrote:
She fallback to user space can happen even during a call? Or is just
about when the call is initialized, the application detects is some
problem when setting up forwarding rules in the kernel and goes for user
space.
It can happen any time.
On 04/10/13 04:17, Klaus Darilion wrote:
I think it is a bad idea to name the relay mediaproxy-ng and the
corresponding Kamailio module rtpproxy-ng.
I've considered that. Apart from the other reasons already mentioned,
for me the deciding factor was that the new module forms a drop-in
On 07/04/13 05:21, Khue Nguyen Minh wrote:
Hi Andreas,
I' using rtpproxy and I can hear voice but quality is not good. After I
changed to mediaproxy-ng, I don't hear anything. I use same config with
rtpproxy. Do you guide change config to mediaproxy-ng?
They're mostly compatible, just make
On 07/07/13 12:28, Juha Heinanen wrote:
why is it that ice relay candidate attributes are added only when
rtpproxy_manage is called, i.e., why not also when
rtpproxy_offer/rtpproxy_answer are called?
Shouldn't make a difference, they should behave the same. Otherwise
please post log excerpts
On 07/08/13 01:52, Juha Heinanen wrote:
i haven't tried it yet. just read what readme says:
4.7. ice_candidate_priority_avp (string)
If specified and if value of the avp value is not 0, rtpproxy_manage
function adds ICE relay candidate attributes to sdp stream(s) containing
ICE candidate
On 07/08/13 08:23, Juha Heinanen wrote:
Richard Fuchs writes:
If you want to use mediaproxy-ng with its ICE features, you'll have to
use it through the rtpproxy-ng module, which is currently available as
patch here:
https://github.com/sipwise/kamailio/blob/master/debian/patches/sipwise/rtproxy
On 07/08/13 09:12, Juha Heinanen wrote:
i read readme of mediaproxy-ng module and don't find the rtpproxy module
capability to tell priority of added ice attributes. also, i have hard
time parsing this these sentences:
+ + - instructs the RTP proxy to discard any ICE attributes
On 07/09/13 08:43, Juha Heinanen wrote:
why are ip4 addresses 0.0.0.0? my mediaproxy-ng is running with these
options and i did not pass ip address as param too offer:
$ ps ax | egrep media
8678 ?Sl10:19 /usr/sbin/mediaproxy-ng --ip=192.98.102.10
--listen-ng=127.0.0.1:25060
On 07/09/13 07:44, Juha Heinanen wrote:
i got so far that my sip proxy started ok and was able to connect to
mediaproxy-ng. i see in syslog these kind of messages:
Jul 9 14:40:11 siika /usr/sbin/sip-proxy[10499]: INFO: rtpproxy-ng
[rtpproxy.c:1410]: rtpp_test(): rtp proxy
On 07/07/13 19:19, Nick Khamis wrote:
Last question from me. Does mediaproxy-ng have the media based
accounting functionality that the original mediaproxy have? We
currently have a problem where we are using RTPProxy along with
CDRTool instead of the MediaProxy because of the lack for NAT
On 07/11/13 13:04, Juha Heinanen wrote:
regarding r flag, if sip ua is behind nat, how can ip address in sdp
be trusted, because source address of rtp packets does not match the
one in sdp?
Mediaproxy-ng pays attention to the source address of incoming packets
and adjusts the forwarding
On 07/12/13 05:48, Juha Heinanen wrote:
it is not what the above description tells. it just tells where
rtpproxy takes the address, nothing about when it starts to send
packets. proper use of rtpproxy or mediaproxy-ng is difficult until
this has been clarified. waiting for both parties to
On 07/12/13 08:32, Richard Fuchs wrote:
On 07/12/13 05:48, Juha Heinanen wrote:
it would result in double rewrite of the sdp when one motivation of
using mediaproxy-ng is that it does the rewriting only once. there
should be possibility pass sdp ip address as parameter in offer and
answer
On 07/14/13 08:59, Alexey Rybalko wrote:
Regarding the mediaproxy-ng documentation special features can't be
invoked without usage of 'ng' protocol provided by rptmediaproxy-ng.
Unfortunately there is no info about rtpproxy-ng module itself. Haven't
found it at Sipwise GitHub site.
You can
Hi,
On 07/18/13 08:48, Alexey Rybalko wrote:
Just suggest someone already tried mediaproxy-ng with conversion
RTP/SRTP. Few examples of options' usage would be very appreciated! May
authors bring them into the tutorial?
E.g. caller invokes RTP/SAVPF profile (SIP over WS), but calle supports
On 07/18/13 17:33, Alexey Rybalko wrote:
Richard,
to be frank, I tried Sipwise's distribution of Kamailio (NGCP 2.8).
Thanks for configured distro image as well :) Spending some time with
tracing the config file brought the call stack: ROUTE_INVITE
-...-ROUTE_BRANCH_ACC_RTP. However I've
On 07/19/13 14:46, Alexey Rybalko wrote:
Good! When NGCP 3.0 will be available for the community?
Have we any chance to evaluate media profile conversion(SDP) prior that
event using base Kamailio? There a several patches from Sipwise for
Kamailio core and some other modules as well (e.g.
On 07/24/13 05:45, Khue Nguyen Minh wrote:
Hi all,
I am using rtpproxy-ng to control mediaproxy-ng. I was install and
config follow this guide:
https://github.com/sipwise/mediaproxy-ng
when I run kamailio with rtpproxy-ng module and mediaproxy-ng I got error:
mediaproxy-ng[25216]: Failed
=0x7fffb6206760, msg=0x7fd617cb13a0,
op=OP_DELETE, flags_str=0x7fd617cd5f90 fox, body_out=0x0) at
rtpproxy.c:1191
Please help me fix it.
Thanks
Khue.
2013/7/24 Richard Fuchs rfu...@sipwise.com
mailto:rfu...@sipwise.com
On 07/24/13 05
On 08/01/13 09:10, Alexey Rybalko wrote:
Hi!
Few days ago I was lucky to establish calls between Chrome and SIP UA.
Thanks to new rtpproxy developers! That was for audio only because many
UAs lack for VP8 support. To verify a video I tried to involve Jitsi
into the tests. Mediaproxy can't
On 08/20/13 10:56, Peter Dunkley wrote:
Hello,
I am testing mediaproxy-ng (running on CentOS 6 on Amazon EC2) for
WebRTC to non-WebRTC calls and I am getting one-way audio most (but not
all) of the time.
I always get audio in the WebRTC to non-WebRTC direction.
Has anybody had any experience
(Posting the follow-up of our off-list discussion for completeness)
On 08/22/13 10:36, Peter Dunkley wrote:
I think (but I need to do more testing) the one way audio was related to
early media and some issues with the media gateway I am using (which was
generating the early media). Removing
Hello,
There was a small code artifact in the rtpproxy-ng module that broke
communication of IPv6 source addresses to mediaproxy-ng. Please update
your Kamailio sources from git master and recompile, that should fix it.
cheers
On 11/11/13 10:29, Pavel Miskov wrote:
Hi all,
I'm trying to
Hey,
You're right, mediaproxy-ng is inconsistent with the docs. Just to
clarify, when you call
rtpproxy_manage(co,10.17.0.102);
you expect the new, rewritten SDP to come out with this IP address as
media address in it, as opposed to the address(es) set on the MP-NG
command line, right?
Right
1, 2014 1:36 AM, Richard Fuchs rfu...@sipwise.com
mailto:rfu...@sipwise.com wrote:
Hey,
You're right, mediaproxy-ng is inconsistent with the docs. Just to
clarify, when you call
rtpproxy_manage(co,10.17.0.102);
you expect the new, rewritten SDP to come out
Hey,
If you're trying to connect two WebRTC endpoints with each, you don't
need any of mediaproxy-ng's magic to get it working. All the previous
replies were assuming that you were trying to connect a WebRTC endpoint
with a non-WebRTC one, which is usually what people are trying to do.
In your
On Wed, Feb 5, 2014 at 5:41 PM, Richard Fuchs rfu...@sipwise.com
mailto:rfu...@sipwise.com wrote:
Hey,
If you're trying to connect two WebRTC endpoints with each, you don't
need any of mediaproxy-ng's magic to get it working. All the previous
replies were assuming
On 02/06/14 14:42, Muhammad Shahzad wrote:
Great, i would test Bundle right away. Just wondering if this branch
also supports DTLS--SRTP. I would love to test that feature when available.
Not quite yet, but it's being implemented as we speak.
cheers
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On 02/12/14 04:41, Mihai Marin wrote:
Hello,
I managed to try it out and I have good news and bad news :)
The good news is that always TURN is working perfect. So, if I remove
all the ice-candidates (rtpproxy_manage(+)) everything is perfect.
That's good to hear!
If
I just append turn
On 02/18/14 11:12, Mihai Marin wrote:
Hello Sirs,
Thank you, one step forward but still buggy - half buggy :)
Now, it's working just one way. If bob calls alice, alice will receive
video but bob won't. If I stop mediaproxy-ng process (without any other
modification) and redo the call,
On 02/20/14 04:15, Mihai Marin wrote:
Hello Sirs, Sir Richard,
I understand the problem but I don't understand the behavior. Let me
tell you how I understood the problem and where I misunderstand the
behavior.
BOB sens an offer to Alice with rtcp-mux. The flow is: UAC (bob) -
Kamailio -
On 02/22/14 07:07, Mihai Marin wrote:
Hello Sirs, Sir Richard,
Thank you for your detailed explication.
I'm still thinking on that but I would say to act as the caller and keep
caller decision. If caller makes an offer with rtcp-mux ,
include separate ICE candidates for RTCP for media proxy
On 02/22/14 07:07, Mihai Marin wrote:
Hello Sirs, Sir Richard,
Thank you for your detailed explication.
I'm still thinking on that but I would say to act as the caller and keep
caller decision. If caller makes an offer with rtcp-mux ,
include separate ICE candidates for RTCP for media proxy
Hey,
Your use case (injecting ICE candidates only) won't work with Firefox
right now, as mediaproxy-ng now speaks DTLS-SRTP and so wants to use its
own DTLS certificate when advertising SRTP. Since FF's certificate won't
match MP-NG's certificate, the DTLS handshake can always only ever work
On 03/26/14 13:42, Mihai Marin wrote:
Hello Sirs, Sir Richard,
To be honest I don't understand why DTLS certificate problem is not
reproducing when overriding ICE candidates (forcing media streams though
MP-NG). In my mind it's should be something similar but without removing
already present
Hey,
Unfortunately your packet dumps are truncated and don't show the
complete SDP bodies. It would also be interesting to see which options
and parameters are passed to mediaproxy-ng when processing the SDP. You
would find this info in the log produced by mediaproxy-ng, which should
also include
://testers.com
2014-04-01 21:41 GMT+03:00 Richard Fuchs rfu...@sipwise.com
mailto:rfu...@sipwise.com:
Hey,
Unfortunately your packet dumps are truncated and don't show the
complete SDP bodies. It would also be interesting to see which options
My guess would be that it's due to a discrepancy between WebRTC and RFC
5764. WebRTC uses a protocol string of RTP/SAVPF, while the RFC says
that UDP/TLS/RTP/SAVPF shall be used. You can try an SDP rewrite
operation to substitute one for the other. Or teach your non-RTC client
to use a different
On 04/03/14 15:32, Olli Heiskanen wrote:
Hello,
Thanks, I'll give that a try and post back. I guess I install and run it
just like mediaproxy-ng?
Yeah, pretty much. Lots of internal changes, but externally the biggest
change is the name.
I'll also try different sip clients like zoiper etc.
On 04/08/14 03:00, Olli Heiskanen wrote:
Hello,
Thanks Juha, that will be a good thing to investigate more when I get my
simple unrealistic scenario working. :)
I tried compiling rtpengine on Centos 6.5, I wonder do I need to change
the Makefile somehow for CentOs? Remove Debian
On 04/10/14 09:26, Olli Heiskanen wrote:
Hello,
After some tests, I'm still having some strange results.
When calling from ws client to grandstream, I get the below output to
/var/log/messages.
In a sip trace after 488 there are only INVITEs from kamailio server to
grandstream but no
On 04/12/14 09:31, Olli Heiskanen wrote:
Hello,
I'm probably still doing something wrong, I still get 488 from the
grandstream. Also zoiper refuses the call with 415 Unsupported Media Type.
According to the module description I tried to change my config to this:
Btw, thanks for enabling
On 04/23/14 11:53, Alex Balashov wrote:
Hello,
I'm running the latest pull of Kamailio 4.1 (last commit
be187e135b0b9b28136817c3569ab5c0abcc5b3f) and am using rtpproxy-ng with
a recent mediaproxy-ng master (commit
cb6990e43864b077dd6a24acfbdf5ef76c1a427e).
For no apparent reason,
On 04/23/14 12:59, Alex Balashov wrote:
The reason I had not previously considered this possibility is because
the documentation says--or, at least to my lackadaisical
interpretation--that rtpproxy_manage() will only call rtpproxy_answer()
if it is operating on a 1xx/2xx reply with SDP,
On 04/23/14 13:24, Alex Balashov wrote:
On 04/23/2014 01:22 PM, Richard Fuchs wrote:
Main selection criterion is whether the message is a request or a
reply, second criterion is the SIP method (taken from the CSeq)
and/or the response code in case of a reply. The route type is only
Hi,
Can you check if the original offer contains an a=setup:actpass
attribute? I remember Firefox having a problem with this in some
version. This attribute is required for DTLS-SRTP and Firefox was not
sending it. It's fixed in later versions.
cheers
On 04/25/14 07:51, Alexey Rybalko wrote:
On 04/26/14 17:32, Alexey Rybalko wrote:
No success for both browsers. It's should be noticed that Chrome
provides both SDES (crypto) and DTLS (fingerprint+setup:actpass)
attibutes (does DTLS have priority in a such case?). However rtpengine
doesn't provide such SRTP data. May be any
On 04/26/14 18:24, Alexey Rybalko wrote:
Failed to set remote answer sdp: Called with a SDP without crypto
enabled (Chrome)
RTPEngine log is attached.
Please try again with ICE=force instead of force_relay, or (more
conservatively) ICE=remove in the offer and ICE=force in the answer. You
On 05/16/14 02:45, Alexey Rybalko wrote:
Hello!
During a call from classical SIP softphone to WebRTC there's no media
from the browser (Mozilla, the same result is for Chrome). In case of a
call from the browser to the softphone there's media flow from both sides.
The snippets from
On 05/16/14 20:30, Alexey Rybalko wrote:
During the call from Fire I saw a lot of SRTP output wanted, but no
crypto suite was negotiated messages from rtpengine. However DTLS is
finally was established. Is that one more issue of Firefox?
Some DTLS-SRTP endpoints seem to be slow with starting
On 05/28/14 13:31, LAA wrote:
Hi all,
I'm currently running a pilot with kamailio 4.1.3 stable, and I would
like to test WebRTC Capabilities. Websockets Support is runnig OK, and
now I'm trying to deal with calls between WebTRC and legacy softphones.
I have installed rtpengine (as it a
On 05/31/14 10:57, Juha Heinanen wrote:
i noticed that there is new entry in ngcp-rtpengine changelog:
ngcp-rtpengine (3.3.0.0+0~mr3.4.0.0) unstable; urgency=low
does that mean that there is new stable release 3.3? what does
~mr3.4.0.0 mean?
The versioning of rtpengine will be carried
On 03/06/14 12:56 PM, Slava Bendersky wrote:
Hello Everyone,
I am trying setup mediappoxy on the gateway and log get fill up with
this message.
Hi,
Use --listen-ng instead of --listen-udp when starting the daemon.
cheers
___
SIP Express Router
On 06/07/14 12:39, Alex Balashov wrote:
Hello,
I'm invoking mediaproxy-ng with rtpproxy_offer/answer(ow) and am
getting a scenario where:
INVITE is forwarded to signalling gateway xxx.xxx.xxx.xxx, which returns
an SDP answer of yyy.yyy.yyy.yyy:51964. rtpproxy is invoked in both
On 06/09/14 10:52, Alexey Rybalko wrote:
Hello to all!
I encountered strange issue with rtpengine: voice during a call is heard
like random binary data. (Video freezes during a video call). _It goes
fine during 2-3 seconds_ before the media flow becomes glitched. It's
related to SRTP calls
On 06/10/14 05:41, Alexey Rybalko wrote:
Hello!
2014-06-09 19:06 GMT+04:00 Richard Fuchs rfu...@sipwise.com
mailto:rfu...@sipwise.com:
Hard to tell what the problem is without looking at the RTP traffic. The
log looks fine. The delay you mentioned could indicate that it might
On 20/06/14 05:31 AM, Alex Balashov wrote:
Thanks for this clarification, Richard. I really appreciate it.
By asymmetric flag, which flag do you mean precisely? I assume 'r'?
It would be the a flag, however I should mention that I was describing
things from an rtpengine perspective, which
On 06/21/14 18:19, Alex Balashov wrote:
Hello,
Despite this fix
https://github.com/sipwise/rtpengine/commit/cbe1f805363b3d6a117e9e5425d79943ddbf92a0
I am continuing to experience periodic crash problems under high loads
with rtpengine. This is after upgrading mediaproxy-ng -
On 07/07/14 06:40 PM, Yuriy Gorlichenko wrote:
Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine.
Kamailio installed as frontend (registration, auth, proxy ) of
asterisksk servers.
WEBRTC users registred at kamailio and asterisk works as media server.
When I try to
On 07/10/14 20:25, Muhammad Shahzad wrote:
Hi,
I am trying to upgrade from mediaproxy-ng to rtpengine trunk version.
The compilation steps go well and i have deb packages created. However
when i try to install them (on same machine where they compiled), i get
this error for every deb
On 07/16/14 03:21, Juha Heinanen wrote:
i build debian packages for rtpengine branch 3.3.0.0+0~mr3.4.1.0. after
that i did 'apt-get dist-upgrade' and it failed like this:
Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ...
Installing new version of config file
On 07/16/14 17:25, Yuriy Gorlichenko wrote:
Hello Rtpengine (rtpproxy-ng module) works fine with kamailio till today.
Without any changes at kamailio or rtpengine kamailio ignores changed by
rtpengine SDP content.
To check this I use sdp_get() and after tying to call I print avp from
this
On 20/07/14 01:15 PM, Olli Heiskanen wrote:
Hi,
...
There may be something off in my Asterisk configs since it's Asterisk
that responds 488, but see how Kamailio responds, SDP contains 2 similar
m= lines. Is there something I might be doing wrong in configuring
rtpengine? The INVITE going to
On 07/23/14 05:03, Olli Heiskanen wrote:
Hi,
Thanks very much for this, that solved the double-m-line issue. Now I'm
calling rtpengine_offer in a branch route.
One issue still remains; the call still gets connected to the called
zoiper client, but it gets hung up right away. I traced
On 24/07/14 09:27 AM, Olli Heiskanen wrote:
That's odd... I pulled a new version from git master 4 days ago, and
copied the compiled rtpengine to /usr/sbin, which is running. (although
might help verifying the version if command rtpengine --version gave
actual output instead of 'undefined') :)
On 04/08/14 01:10 PM, Paul Belanger wrote:
Greetings,
I'm having some trouble getting dtls-srtp working with kamailio 4.1
(mediaproxy-ng) and rtpengine (master).
I believe I finally have my branch logic setup properly in kamailio,
however when the calls get placed into rtpengine, it appears
On 05/08/14 11:01 AM, Paul Belanger wrote:
I was hoping somebody could confirm the following is a _normal_ log
file for rtpengine. Specifically I am curious of the 'Successful STUN
binding request from' messages are continuously logged.
Chrome seems to be doing this during normal operation. I
On 08/11/14 17:04, Narsay, Deep wrote:
Hello Andreas,
Yes, that's what I had thought, but
I am actually seeing it using two different UDP ports towards Freeswitch
(send=40036 and recv=40042)
and two more ports towards SIP Client (send=40038, and recv=40040).
I am wondering if there
On 08/19/14 14:53, Anthony Messina wrote:
I've begun playing with rtpengine (git e0957d1) a little and in my testing:
CSipSimple - Kamailio/rtpengine - Asterisk 12/13
I see the following error when parsing SDP from the Asterisk side, possibly
related to the use of the FQDN in the Origin:
On 08/26/14 20:58, Alex Balashov wrote:
On 08/26/2014 08:56 PM, Paul Belanger wrote:
I'd agree 'drop-in' replacement is not correct. I ran into the same
issues as you. Current there is no bridge-mode in rtpengine, I point
you to an open issue about it [1].
I think the idea behind the
On 08/25/14 19:25, Alex Villacís Lasso wrote:
I have a rtpproxy configuration that spawns several rtpproxy instances,
using bridge mode. An example is shown below:
/usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s
udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2
On 09/04/14 12:01, Daniel Tryba wrote:
On Thursday 04 September 2014 15:22:52 Daniel-Constantin Mierla wrote:
- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp1673992
rtpengine (former rtpproxy-ng) should have it as well, I guess.
Found this before posting, but I could not
On 08/25/14 19:25, Alex Villacís Lasso wrote:
I have a rtpproxy configuration that spawns several rtpproxy instances,
using bridge mode. An example is shown below:
/usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s
udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2
On 08/21/14 03:09, Juha Heinanen wrote:
Now I made another dist-upgrade upgrading rtpengine from
3.3.0.0+0~mr3.4.1.0 to 3.3.0.0+0~mr3.5.0. Indeed dist-upgrade tries to
setup daemon after the kernel module was removed but before new kernel
module was installed.
...
Setting up
On 09/24/14 09:16, Sebastian Damm wrote:
Hi,
I switched from rtpproxy module to the rtpproxy-ng module lately, and
noticed a strange behavior. In my branch route to the device, I have two
statements:
fix_nated_sdp(1);
rtpproxy_offer();
The first command appends a line with
On 09/25/14 10:22, Frank Carmickle wrote:
On Sep 25, 2014, at 10:09 AM, Marino Mileti marino.mil...@alice.it
mailto:marino.mil...@alice.it wrote:
Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would
like to reach all of them in parallel mode. I can't use for all of
On 09/29/14 13:03, Richard Fuchs wrote:
On 09/25/14 10:22, Frank Carmickle wrote:
On Sep 25, 2014, at 10:09 AM, Marino Mileti marino.mil...@alice.it
mailto:marino.mil...@alice.it wrote:
Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I
would like to reach all of them
On 09/25/14 10:41, Marino Mileti wrote:
No no. The video will be sent by the caller user to all the callees.
I'l try to explain better. My scenario is:
- A make a call to a group... B C are group member...so Kamailio is
able to call them in parallel using alias..
- B C receive the
On 09/25/14 12:05, Jeff Pyle wrote:
Hello,
Given the following scenario with Kamailio and rtpengine in the middle:
- call establishes with G.711 RTP
- b-leg re-invites to T.38, indicating a different port number then he
is using for G.711
- a-leg refuses the re-invite with a 488
-
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(rtpproxy, extra_id_pv,
On 09/29/14 13:19, Frank Carmickle wrote:
On Sep 29, 2014, at 1:14 PM, Richard Fuchs rfu...@sipwise.com wrote:
This may work with rtpengine, as it will open new ports for answers come
from different endpoints. But the final two-way association for the
actual call may still end up broken
On 09/29/14 13:29, Frank Carmickle wrote:
On Sep 29, 2014, at 1:24 PM, Richard Fuchs rfu...@sipwise.com wrote:
On 09/29/14 13:19, Frank Carmickle wrote:
On Sep 29, 2014, at 1:14 PM, Richard Fuchs rfu...@sipwise.com wrote:
This may work with rtpengine, as it will open new ports
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 umbrella
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 14:23, Marino Mileti wrote:
The problem isn't on 183s but on the multiple INVITE that Kamailio sends to
clients behind rtpengine. Rtpengine open new ports for answer but on INVITE
the rtpengine ports are the same...This happens because for all these
clients the callid is still the
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