17 Sep
2014
17 Sep
'14
4:15 p.m.
Hi Daniel,
Here is something i traced in the log:
ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force'
ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general
protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]
What's the cause of this error? i am using code from the master branch.
Perhaps this has something to do with the rptengine service
crash/termination.
Regards
On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini <
abhishek.saini(a)enukesoftware.com> wrote:
Hi Daniel,
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.
I also updated this param : modparam("nathelper", "sipping_from",
"
sip:pinger@abc.com") to my domain
Now the scenario is as follows:
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.
2) Still when i call from webrtc to iphone - the retpengine service of
ubuntu terminates/crashes (like before) and needs to be restarted.
Does it have any thing to do with rtp port ranges? or is there some other
misconfiguration?
Regards,
Abhishek
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
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.
Cheers,
Daniel
On 16/09/14 14:24, Abhishek Saini wrote:
Hi Daniel,
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.
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.
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)
Following are the links to my latest kamailio.cfg file and port trace
log of sip messages.
http://jmp.sh/o0apKgP
http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards,
Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini <
abhishek.saini(a)enukesoftware.com> wrote:
Hi Daniel,
Thanks for this.
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:
rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it
enabled
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
ERROR: <script>: ==> duri=[
sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp]
INFO: <script>: Request coming from WS
ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown
option ` '
INFO: <script>: Reply from softphone: 100
And this SIP message:
SIP/2.0 603 Failed to get local SDP.
Regards,
Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Hello,
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.
Searching on web, I see that Carlos has published a config for it, see:
- https://github.com/caruizdiaz/kamailio-ws
Cheers,
Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and
mediaproxy-ng package on my ubuntu box. As suggested here:
http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
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:
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/TCP
54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
Via: SIP/2.0/WS
df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
From: "admin" <sip:admin@abc.com>;tag=bzhwwG8nT2gFwwJgIyrz.
To: <sip:hari@abc.com>;tag=OIllTQf.
Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
CSeq: 65463 INVITE.
User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
Supported: replaces, outbound.
Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards,
Abhishek
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany