Hi Bogdan,
Thanks for your support most of the system is stable if i dont use the NAT i.e
directly used the public IP over opensips machine.
few problem running opensips (version 2.2) behind NAT (might be different RTP
source ) i modified the below options in cfg file
advertised_address = "49.255.xx.xx" advertised_port = 5060
rtpproxy_offer("co","49.255.xxx.xx");
following the below links
https://blog.opensips.org/2017/10/25/running-opensips-in-the-cloud
But after this change my RTP proxy got crashed i need to restart that again and
again for every call ( with no voice)
Aug 14 22:16:40 localhost opensips: Aug 10 22:16:40 [15887]
DBG:rtpproxy:force_rtp_proxy: force rtp proxy with param1 <co> and param2
<49.255.xxx.xx>Aug 14 22:16:40 localhost opensips: Aug 10 22:16:40 [15887]
DBG:rtpproxy:force_rtp_proxy: Forcing body:Aug 14 22:17:16 localhost rtpproxy:
INFO:handle_delete:Z6UzGDogqEpan1rHLieTKw..: forcefully deleting session 1 on
ports 25880/0
Aug 14 22:17:16 localhost rtpproxy:
INFO:remove_session:Z6UzGDogqEpan1rHLieTKw..: RTP stats: 0 in from callee, 0 in
from caller, 0 relayed, 0 droppedAug 14 22:17:16 localhost rtpproxy:
INFO:remove_session:Z6UzGDogqEpan1rHLieTKw..: RTCP stats: 0 in from callee, 0
in from caller, 0 relayed, 0 droppedAug 14 22:17:16 localhost rtpproxy:
INFO:remove_session:Z6UzGDogqEpan1rHLieTKw..: session on ports 25880/0 is
cleaned up
I downloaded RTP from below link (suggest if this is the correct source )
https://github.com/sippy/rtpproxyyyy
as some one suggest me to download and use from source Index of /pub/rtpproxyy
Please suggest which is correct source.
ThanksSanjeev!!
On Thursday, 9 August, 2018, 7:35:06 PM GMT+10, Bogdan-Andrei Iancu
<bog...@opensips.org> wrote:
Hi Sanjeev,
Well, things are getting a bit more complex here as you run OpenSIPS behind a
NAT (in a private network) - this means that opensips will be 'visible' with
different IPs by the parties in the same private network and by the parties in
the public network.
In the same time you need to ensure that RTP is able to be routed (at IP
level) between the 2 endpoints - probably that's your issue, that the private
IP advertised in SDP by the end point in the private network is not routable
from the perspective of the other end point in the public network -> the public
end point cannot send RTP to the private endpoint.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/03/2018 10:10 AM, Sanjeev Sharma wrote:
Hi ALL
as a further dig into the media issue no voice between caller and callee and
found that the in the request packet of invite header the value of via header
is coming incorrect ( as in the below packet is coming the local ip address of
the machine 192.1682.248 instead of 159.200.37.234.
my machine is behind the fortigate firewall. could you please suggest how to
set or change the value of via header , is this will be done at firewall level
or to change the any value in the opensip configuration file. below is sample
packet
U 2018/08/03 16:27:25.156824 159.200.37.234:5060 -> 149.355.453.253:5060
INVITE sip:124007@149.355.453.253;transport=UDP SIP/2.0. Via: SIP/2.0/UDP
192.168.2.248:5060;branch=z9hG4bK-524287-1---81e029059dd869e2;rport.
Max-Forwards: 70. Contact: <sip:124010@159.200.37.234:5060;transport=UDP>. To:
<sip:124007@149.355.453.253;transport=UDP>. From:
<sip:124010@149.355.453.253;transport=UDP>;tag=29295014. Call-ID:
0PqYYFAWxnuGearNQ73LxQ... CSeq: 1 INVITE. Content-Type: application/sdp.
User-Agent: Z 3.15.40006 rv2.8.20. Allow-Events: presence, kpml, talk.
Content-Length: 241.
Thanks in advance Sanjeev!! On Wednesday, 1 August, 2018, 5:41:02 PM
GMT+10, Sanjeev Sharma <sanjeevt...@yahoo.com> wrote:
Hi Bogdan-Andrei, Thanks for the response! it motivated me to read more
stuffs related to opensips. I followed the steps of installation steps and
type of route define in https://www.opensips.org/ but again i am facing
problem in my first installation of stepup Earlier the opensip server was
behind the NAT - Public address on firewall and opensip machine (Centos) having
a local address (192.168.2.x) , but now the opensips machine (version 2.2)
directly hosting public address without mapping ( Lan cable from fortigate
firewall to opensips machine with any mapping or port block all ports open)
Scenario is If my UAC1 (zoiper on my laptop1) , UAC2 (zoiper on my laptop2 )
and Opensip machine are in the same network i.e ISP1 i can easily hear the
voice / audio between the UAC1 and UAC2 #) But i change the Network of UAC1
(zoiper on my laptop1) , UAC2 (zoiper on my laptop2 ) to connect with ISP2
and opensip machine remain on ISP1 then i am able to register and call UAC but
their is no voice / audio ( RTP Media) among the user agent. ( i am 2 ISP
network from different provider) I the last 1 week i tried all the solution
what ever i am able to find online but still its does work. could you please
suggest how to troubleshoot further. additionally is their any repository where
i can study more about the route how to they work and how to change / Set the
the value in header field of the request / response. guidance and direction at
stage will help me move further Thanks in advance , i know few are my quires
are wired as i just enter in the world of opensips Thanks Sanjeev Kr Sharma
On Tuesday, 24 July, 2018, 8:11:16 PM GMT+10, Bogdan-Andrei Iancu
<bog...@opensips.org> wrote:
Hi Sanjeev,
As I understand correctly, you end up connecting into your opensips devices
from different networks - a devices from the private network (same as opensips)
and another device from the public network.
But note that bidirectional direct communication between the 2 devices is not
possible, as the public device cannot send traffic to a private IP/destination.
Depending on the opensips cfg, the SIP signaling works, as OpenSIPS will act
as a bridge between the 2 networks. But this is not true for RTP, as RTP goes
directly between the 2 devices. So, what you need is to use a media relay
acting a bridge between the 2 networks.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 07/24/2018 05:51 AM, Sanjeev Sharma via Users wrote:
Hi ,
i am new to opensips world and i installed the openSIPS version 2.2 , facing
issue of media. Setup is my configuration is like 1) UAC request come through
fortigate firewall on public address (1234@49.121.121.121) and then passed to
my opensip machine (centos 7 , opensip version 2.2) having local address (
natted with public address) 2) registration of UAC is fine and if UAC are on
same network lets say one client on my laptop and another on mobile device
(both on same ISP WIFI) then voice is going through between the UAC but i
switch the network of the mobile to telco service provider then their is no
media pass on in between the UA or some time only one UA able to listen the
other side.
i looked online and search lots of stuffs related to this and changed in
configuration but unable to solve or find what and where i am getting. i tried
TCP dump for both kinds of call i.e having voice or no voice but unable to
identify the difference between calls having voice and no voice.
Since i am new to this setup and configuration , just stuck due to above
problem .
Please suggest where to look and what could be the possible reason. Currently
all traffic ( i.e all ports open) being allow from firewall to machine
Thanks Sanjeev
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