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:[email protected];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:[email protected]:5060;transport=UDP>.
To: <sip:[email protected];transport=UDP>.
From: <sip:[email protected];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
<[email protected]> 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 inhttps://www.opensips.org/
<https://www.opensips.org/Documentation/Configure-File-2-2>
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
<[email protected]> 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
([email protected] <mailto:[email protected]>) 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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