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 check
RTP-Proxy
>
> Yes. Ipv4/Ipv6 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
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
Hi,
While I can't really answer your question, the logic in mediaproxy-ng is
that if the to-tag is given in the "D"elete message, it has to match the
to-tag that was previously given in the "L"ookup message alongside with
the from-tag. If no to-tag is given in the delete message, then only the
fro
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
On 10/22/12 13:59, Juha Heinanen wrote:
> Daniel-Constantin Mierla writes:
>
>> For safety, i would use 48, to allow zero termination
>
> why 48 when max length of ipv6 addr is 39 chars? did you mean 40?
If you use 4-in-6 mapped format, omit zeroes compression and add square
brackets, you get 4
On 10/22/12 14:14, Juha Heinanen wrote:
> Richard Fuchs writes:
>
>> If you use 4-in-6 mapped format, omit zeroes compression and add square
>> brackets, you get 47 characters:
>>
>> [::::::123.123.123.123]
>
> why would someone use s
On 10/22/12 14:19, Juha Heinanen wrote:
> to me it makes sense to store addresses only in canonical format to
> database in order to make sure that address comparison can be made
> uniquely.
Using the string representation of an address for anything other than to
provide a human readable version
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.
>>
>
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
>
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
> tutorial/documen
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 o
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 (a
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 tim
Hello,
On 04/09/13 12:19, Iñaki Baz Castillo wrote:
> So the "received" value added by Kamailio is invalid. Such a value
> cannot be the value of a SIP URI parameter at all. I strongly propose
> encoding it in base64 or escaping the "=" and the ";" symbols when in
> a SIP URI param value as Juha
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
rep
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 sur
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 w
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 attr
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/rt
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=127.0.0.1:2
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 found,
> s
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 s
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
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 p
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 c
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
RT
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
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]: Fa
15#012a=fmtp:123
>>> 124/124/124/124#015#012a=pcfg:1
>>>
t=1#015#012a=sendrecv#015#012a=rtcp:40005#015#012a=ice-ufrag:4SUsQrLE#015#012a=ice-pwd:qhLmZW7KFb6W7QOVFEOraZfcQaCG#015#012a=candidate:rfWFb7Vp3QisWSvf
>>> 1 UDP 2130706432 0.0.0.0 40004 typ
On 07/28/13 22:37, Khue Nguyen Minh wrote:
> Hi Richard,
>
> I get mediaproxy from link:
> https://github.com/sipwise/mediaproxy-ng
>
> and rtpproxy-ng from link:
> https://github.com/sipwise/kamailio/tree/3.3+ngcp2.8
And how did you download them from there? Through a regular git
clone/checkou
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 rewri
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 o
(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 th
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?
Ri
ne, rtpproxy.
>
> On Feb 1, 2014 1:36 AM, "Richard Fuchs" <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&
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 ca
`a=ice-ufrag:gNml+vA5NqfaRg0w
> a=ice-pwd:dQJW2XWJ+g6gTIujfT819g2d
> a=ice-options:google-ice
> a=mid:audio
> ...........
> 14(21473) DEBUG: [parser/sdp/sdp.c:574]: parse_sdp_session():
> ignoring unknown type in a= line: `a=ssrc:320
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 09/01/16 07:28 PM, Arsen Hovhanissian wrote:
Hi everyone, i’ve been having this “not so problem” going on. So I have
rtpengine installed on a server and use the default rtpproxy module on kamailio
and it works beautifully. Having read the rtpengine modules description, I see
that it is a d
On 10/01/16 12:48 AM, Arsen Hovhanissian wrote:
Hi Richard! Here’s the output at log level 7
[1452404737.610730] WARNING: Failed to properly parse UDP command line '11683_0
d7:command4:pinge' from 10.0.0.10:54602, using fallback RE
[1452404737.619006] WARNING: Failed to properly parse UDP comma
On 01/12/2016 04:09 AM, riko nir wrote:
Hi all,
I have a media server and it is able to handle SRTP, provided the crypto
key.
We are planning to give webrtc support to the media server. We are using
opensips+rtpengine for that.
For dtls, we are using rtpengine. The rtpengine just needs to do t
On 01/13/2016 02:37 AM, riko nir wrote:
Hi, Thanks for the answer.
Do you have any options for sending this keys to opensips somehow, by
modifying the code in rtpengine and in opesips script file?
I don't know much about Opensips and so can't provide guidance about how
to pass these values ba
On 01/13/2016 04:26 AM, riko nir wrote:
Hello,
A call from a remote webrtc client is coming to (opensips+rtpengine).
The media streams from the webrtc client is multiplexed. Can I use
rtpengine to demultiplex the multiplexed streams and send it to other
end as de-multiplexed SRTP traffic . This
On 05/04/16 08:03 AM, Juha Heinanen wrote:
I noticed that 15 days ago there was rtpengine commit
https://github.com/sipwise/rtpengine/commit/aac8899b612dc8103b89f3f9c921f88af3501303
titled "Release new version 4.4.0.0+0~mr4.4.0.0", but I could not find
the corresponding branch or tag.
If someo
On 04/05/16 11:16 AM, Yasin CANER wrote:
Hello;
i tried to make rtpengine.so deamon that gives error for
evthread_use_pthreads is implicit.
You need to have a recent libevent installed - libevent-devel or
libevent2-devel.
http://rpm.pbone.net/index.php3/stat/3/srodzaj/1/search/libevent2-
On 05/06/2016 07:36 AM, Dmitry wrote:
> hello
>
> i read the documentation about RTPengine.
> and the documentation says that:
>
> If INVITE with SDP, when the tm module is loaded, mark transaction with
> internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
> rtpengine_answer()
>
> wh
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the
sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/
//
* /Repacketization or transcoding/
This refers to trans
On 05/19/2016 04:52 AM, Moacir Ferreira wrote:
...
> So the Grandstream offers a lot of codecs but will get a "Not Found"
> from Kamailio. Look in the other way:
That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a
missing registration.
> Here the Grandstream says "Media type
On 06/01/2016 04:26 AM, Serge S. Yuriev wrote:
Hello,
Thanks a lot.
If I understand docs correctly there is no such thing for RTPEngine and
we should use app params?
Yes there is, you can give it as "media-address=..." inside the options
string to the respective function calls.
Cheers
On 17/06/16 03:46 AM, Dmitry wrote:
Hi all
I have the following code:
if($T_reply_code=="200")
{
if(has_body("application/sdp"))
{
xlog("L_INFO", "RTPENGINE received internal reply
$T_reply_code $rr SDP extra lines will be removed");
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
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) -
> Kamaili
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 pro
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 pro
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
again
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 pres
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
packet
> from 127.0.0.1:34407 <http://127.0.0.1:34407>: Unknown call-id
> [d3:sdp216:v=0#015#012o=gsclient 8000 8002 IN IP4
> 192.168.0.106#015#012s=SIP Call#015#012c=IN IP4 192.168.0.106#015#012t=0
> 0#015#012m=audio 5030 RTP/AVP 0 13#015#012a=sendrecv#015#012a=rtpmap:0
> PCMU/8000#015#012a=ptime:20#015#012m=audio 0 R
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 differen
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
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
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 enab
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, oth
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
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. Yo
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 star
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 carri
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 (SER
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 c
On 06/10/14 05:41, Alexey Rybalko wrote:
> Hello!
>
> 2014-06-09 19:06 GMT+04:00 Richard Fuchs <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
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, whic
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 cal
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 p
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 /etc/init.d/ngcp-rt
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
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 th
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 tr
On 07/23/14 11:01, Olli Heiskanen wrote:
>
> Thanks,
>
> I think here's all of the call from before the called party answers:
...
I can't seem to reproduce this, when I run through the same sequence,
the answer comes out as a=setup:active as it should. Are you sure you're
using the latest vers
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 DTL
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
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