What is the value set to localnet and externip parameters? NAT will not work
without setting these parameters. These are global parameters.
Regards,
Amit Patkar___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.si
What I found that when sip client registered, it was somehow changed to my
lan address. Hence the audio rtp stream does not go out to the internet
back to sip client.
What should I change?
On Sun, Jan 4, 2015 at 8:39 PM, CK Lee wrote:
> I have this already in my sip.conf
>
> [kamailio-ns]
> typ
I have this already in my sip.conf
[kamailio-ns]
type=friend
host=sip.example.org
port=5060
disallow=all
allow=gsm
allow=g729
allow=alaw
allow=ulaw
context=default
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
qualify=yes
nat=yes
I use asterisk 11 and the setup is ok without any one way au
Please check sip.conf. You need to enable NAT options. Asterisk need to publish
public IP in SDP for RTP traffic to reach your network. Asterisk need to
differentiate local clients and external clients. localnet and externip
parameters should be configured correctly.
Regards,
Amit Patkar
On J
I am new to Kamailio after starting to play around for 3 weeks.
Before Kamailio, I used asterisk and it works quite well in my network
which has 2 public IP addresses and 10+ wired and wireless users. The
asterisk listens to only one IP which my netgear router port forward the
necessary sip and rd
Hi,
I have reported a bug on rptengine github, for the crash issue:
https://github.com/sipwise/rtpengine/issues/27
You mentioned that you have been using rtpengine kamailio module and the
rtpengine debian package with success. Was it on ubuntu box or some other
linux system? Sorry for asking this
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[40+3]
What's the cause of this error? i
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:pin...@abc.com") to my domain
No
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
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
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 found, support for it enabled
ERROR:
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 publi
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 ca
Hi,
It appears that my last two messages have gone in moderation. Anyways, Can
you please tell me, how can i setup rtpengine on Ubuntu machine? After
installation - What configurations will i have to change?
I have lurked the internet a lot but did not find any tutorial on this.
Would appreciate
Hi,
I have not heard on my last reply (it went in moderation). So, I am posting
one ngrep result here, Please let me know on this:
interface: any
filter: (ip or ip6) and ( port 5060 )
#
T 2014/09/04 12:51:26.423430 182.64.39.131:3207 -> 172.31.47.138:5060 [AP]
INVITE sip:h...@abc.com SIP/2.0.
Via
Hi,
Please find attached the output of ngrep for three type of
combinations/connections:
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful
webrtc2blink_same_ntw_failed
webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone
Hello,
maybe you can send to mailing list the output of ngrep so we can look
and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use
rtpengine.
Cheers,
Daniel
On 04/09/14 13:01, Abhishek Saini wrote:
Hi Daniel,
Thanks, i was able to use the co
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the
chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by
calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and h
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the
comment). But it still does not seem to work.
I don't quite know how can i debug,
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the comment). But
it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
Regards,
Abhishek
Hello,
no time to look at config, but if you run the sip server on a private IP
behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video
calls seem to work well when both the devices are connected to the same
network, however, when one device connects to a different network (the two
devices now are on different networks), they are able to register on SIP
This is webrtc, using Kamailio with websocket relay to Asterisk.
I am not using rtpproxy
07.04.2014, 22:49, "Kelvin Chua" :
> is this webrtc?are you using rtpproxy?
>
> Kelvin Chua
>
> On Mon, Apr 7, 2014 at 6:37 AM, jaflong jaflong wrote:
>> Hi,
>>
>> I am at the point where connection is est
is this webrtc?
are you using rtpproxy?
Kelvin Chua
On Mon, Apr 7, 2014 at 6:37 AM, jaflong jaflong wrote:
> Hi,
>
> I am at the point where connection is established and no apparent errors
> are reported.
>
> However audio is not output.
>
> The rtp traffic seems to be transfering between the
Hi,
I am at the point where connection is established and no apparent errors are
reported.
However audio is not output.
The rtp traffic seems to be transfering between the points as conclueded
because Asterisk debug log shows
Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq
Hi List,
I am using dispatcher module to route everything from registerations to
calls toa asterisk pbx. thats all what openser is doing at the moment.
outgoing calls are fine but I get CHANUNAVAILABLE status for incoming calls
on Asterisk. I guess the reverse sip dialogue are not working properly
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