<div dir="ltr"><div><div><div><div><div><div><div>Hi Daniel,<br><br></div>As you instructed, i installed kamailio from the master branch (which has rtpengine module). Along with this, i installed the rtpengine package from sipwise, as instructed by them.<br><br></div>I also updated this param : modparam("nathelper", "sipping_from", "<a href="mailto:sip%3Apinger@abc.com">sip:pinger@abc.com</a>") to my domain<br><br></div>Now the scenario is as follows:<br><br></div>1) I am able to call webrtc(firefox and chrome) from iphone, the signalling seems to be working fine, call can be paused, resumed etc.., but there is no audio/video transmission.<br><br></div>2) Still when i call from webrtc to iphone - the retpengine service of ubuntu terminates/crashes (like before) and needs to be restarted.<br><br></div>Does it have any thing to do with rtp port ranges? or is there some other misconfiguration?<br><br><br>Regards,<br></div>Abhishek<br><div><div><div><div><div><div><div><div><div><div><br> <br><div><br></div></div></div></div></div></div></div></div></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Hello,<br>
<br>
maybe you should play with kamailio master branch (which is in
testing phase before becoming 4.2) -- there you have the rtpengine
-- and see if you get it working. Once that, you can look at using
an older version, knowing you have it working and be able to
compare. As I needed latest features, whenever I needed webrtc
gatewaying, I used devel branch of rtpengine module.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 16/09/14 14:24, Abhishek Saini
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>
<div>Hi Daniel,<br>
<br>
<br>
</div>
I was able to solve a fraction of my problem, Actually, the
github link had used rtpengine.so and i was using
rptproxy-ng.so, there is a difference in the flag
conventions between the two; i modified that to achieve a
little progress.<br>
<br>
</div>
Now, i am able to call on webrtc(firefox) from sip phone.
However, after accepting call, there is no audio, and
disconnecting the call from either end does not disconnect the
call. <br>
<br>
</div>
When i try to call from webrtc(firefox) to sip phone, there is
no signalling at all, and the sip phone to webrtc calls can't
connect after that. (I analyzed that mediaproxy-ng/rtpengine
process terminates and has to be started again)<br>
<div>
<div>
<div>
<div><br>
</div>
<div>Following are the links to my latest kamailio.cfg
file and port trace log of sip messages.<br>
<a href="http://jmp.sh/o0apKgP" target="_blank">http://jmp.sh/o0apKgP</a><br>
<a href="http://jmp.sh/HXnFRQj" target="_blank">http://jmp.sh/HXnFRQj</a><br>
<br>
</div>
<div>I am clueless at the moment!<br>
</div>
<div><br>
</div>
<div>Regards,<br>
</div>
<div>Abhishek<br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Sep 16, 2014 at 1:15 PM,
Abhishek Saini <span dir="ltr"><<a href="mailto:abhishek.saini@enukesoftware.com" target="_blank">abhishek.saini@enukesoftware.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>
<div>
<div>
<div>
<div>Hi Daniel,<br>
<br>
</div>
Thanks for this.<br>
<br>
</div>
I took the entire config files and configured it as
per my ips and ports, after doing that, still no
call establishment(webrtc to classic sip phones and
vice-versa). Following is what i get in
kamailio.log:<br>
<br>
rtpp_test(): rtp proxy <udp:<a href="http://127.0.0.1:7722" target="_blank">127.0.0.1:7722</a>>
found, support for it enabled<br>
ERROR: rtpproxy-ng [rtpproxy.c:1254]:
rtpp_function_call(): unknown option ` '<br>
ERROR: <script>: ==>
duri=[<a>sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp</a>]<br>
INFO: <script>: Request coming from WS<br>
ERROR: rtpproxy-ng [rtpproxy.c:1254]:
rtpp_function_call(): unknown option ` '<br>
INFO: <script>: Reply from softphone: 100<br>
<br>
</div>
And this SIP message:<br>
SIP/2.0 603 Failed to get local SDP.<br>
</div>
<div><br>
</div>
Regards,<br>
</div>
Abhishek<br>
<div>
<div>
<div><br>
<br>
<br>
</div>
</div>
</div>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Sep 15, 2014 at 6:19
PM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Hello,<br>
<br>
the reply code indicates that the media type is
not supported, thus there has been no gatewaying
between webrtc and classic rtp. Just replacing
rtpproxy with rtpengine is not enough, there are
different parameters that have to be provided.<br>
<br>
Searching on web, I see that Carlos has
published a config for it, see:<br>
- <a href="https://github.com/caruizdiaz/kamailio-ws" target="_blank">https://github.com/caruizdiaz/kamailio-ws</a><br>
<br>
Cheers,<br>
Daniel
<div>
<div><br>
<br>
<div>On 15/09/14 12:58, Abhishek Saini
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>Hi, <br>
<br>
</div>
I have successfully setup rtpproxy-ng
kamailio module and mediaproxy-ng
package on my ubuntu box. As suggested
here:<br>
<a href="http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html" target="_blank">http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html</a><br>
</div>
<div><br>
</div>
<div>I have kept rtpproxy-ng's
configuration same as the rtpproxy
module, but still not able to connect
the webrtc calls to classic sip phones
(and vice-versa). Below is the sip
message that is traced:<br>
<br>
<br>
SIP/2.0 488 Not acceptable here.<br>
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$<br>
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$<br>
From: "admin" <<a href="mailto:sip%3Aadmin@abc.com" target="_blank">sip:admin@abc.com</a>>;tag=bzhwwG8nT2gFwwJgIyrz.<br>
To: <<a href="mailto:sip%3Ahari@abc.com" target="_blank">sip:hari@abc.com</a>>;tag=OIllTQf.<br>
Call-ID:
31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.<br>
CSeq: 65463 INVITE.<br>
User-Agent: LinphoneIPhone/2.2.1
(belle-sip/1.3.2).<br>
Supported: replaces, outbound.<br>
Content-Length: 0.<br>
<br>
</div>
<div>Can you please let me know, what's
going wrong and how can i proceed.<br>
<br>
</div>
<div>Regards,<br>
</div>
<div>Abhishek<br>
</div>
<div><br>
<br>
</div>
<div><br>
<br>
</div>
</div>
</blockquote>
<br>
</div>
</div>
<span>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
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</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
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</blockquote></div><br></div>