[OpenSIPS-Users] No Voice Comm in Conference call

Mon Jun 18 07:54:56 CEST 2012

Hi Ian, 

Thank you very much for your reply. 

Sorry for the late response, I had been sick for sometime. 

I checked for what you had asked me and I found, 

1. The SDP packets are not being rewritten with the advertised address. 
2. The RTP Proxy is directly with an Public IP and the is no Firewall needed for it. 

I am still facing the same issue. I am attaching my opensips.cfg file. 

I am newbie to opensips and desperate to get over this issue, hence please let me know where am I making the mistake. 

Thanks, 
Arjun 

----- Original Message ----- 
From: "Ian Buckner" <ian.buckner1 at googlemail.com> 
To: "OpenSIPS users mailling list" <users at lists.opensips.org> 
Sent: Monday, June 11, 2012 4:22:50 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi 
Subject: Re: [OpenSIPS-Users] No Voice Comm in Conference call 

Hi Arjun, 

its the same patch. If it didnt work then the questions I would ask would be: 

1) Is the SDP in the invites being rewritten with the advertised address correctly 
2) Is the Firewall configured to pass the RTP packets across to the internal address rtpproxy is listening on? 

I would check these by taking network traces on the rtpproxy box for 2) and on the opensips box for 1) (I run on Ubuntu so use tcpdump) 

There were no changes needed in the opensips config file needed for me - its just that rtpproxy is returning the advertised address rather than its internal listening address. 

For reference, my rtpproxy startup line looks like this: 

rtpproxy -u rtpproxy -l 192.168.0.1 -s udp:192.168.0.1:6050 -m 6000 -M 6010 -A 89.21.23.237 

So the machine has internal IP address 192.168.0.1, receives OpenSips Mgmt commands on port 6050, uses ports 6000-6010 for RTP sessions and returns a public IP of 89.21.23.237 to OpenSips for SDP. 

Hope this helps, 

Ian 

On 11 Jun 2012, at 11:42, Arjun Shankar K S wrote: 

Hi Ian, 

Please find attached the patch available at my side. 

Thanks, 
Arjun 

----- Original Message ----- 
From: "Arjun Shankar K S" < arjun_ks at kalycito.com > 
To: "OpenSIPS users mailling list" < users at lists.opensips.org > 
Cc: "OpenSIPS users mailling list" < users at lists.opensips.org > 
Sent: Monday, June 11, 2012 4:11:25 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi 
Subject: Re: [OpenSIPS-Users] No Voice Comm in Conference call 

Hi Ian, 

Sincere thanks for your reply. 

I am attaching a patch herewith. Please let me know if you are referring to the same and this patch did not work for me. If it works for you can you pl let me know wat are changes that has to be made in the opensip,cfg file? 

Else if patch is different, pl send me the patch. 

Regards, 
Arjun 

----- Original Message ----- 
From: "Ian Buckner" < ian.buckner1 at googlemail.com > 
To: "OpenSIPS users mailling list" < users at lists.opensips.org > 
Sent: Monday, June 11, 2012 3:48:40 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi 
Subject: Re: [OpenSIPS-Users] No Voice Comm in Conference call 

Hi Arjun, 

Regarding rtpproxy - if it makes your life easier, there is a very simple patch for the rtpproxy source which allows you to specify an advertised IP address on the command line this will be returned to OpenSips and inserted into SDP rather than the listening address. I used this last week for a server behind NAT and it worked perfectly. 

If you want to mail me I'll happily send you a patched tarball of source. 

best, 

Ian 

On 11 Jun 2012, at 11:14, Bogdan-Andrei Iancu wrote: 

Hi Arjun, 

If you have no audio at all for the call to Conf Server, you need to check the signaling, particularity speaking the SDP part to be sure you inserted RTPProxy correctly between UAC and Conf Server. 

So, make a SIP capture for the call to Conf and check if the IPs in SDP are correct. 

Regards, 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com 
On 06/07/2012 05:46 PM, Arjun Shankar K S wrote: 

Hi All, 

Greetings to everyone !!! 

I have set up opensips and RTP Proxy in two different hosts since I have opensips in a Natted environment where RTP Proxy refused to budge. 

Now I have installed RTP Proxy in a direct public IP. Normal calls between 2 Client is working great !! 

During conference call, the calls get connected but there is no voice communication between any of them and soon the client who was connected last, gets disconnected. 

I could not find much support regarding this issue. Any support is sincerely appreciated. 

In my opensips.cfg, I have made the following config changes for RTP in different host, 

------Nat Params------------- 
modparam("usrloc","nat_bflag", 6) 
modparam("nathelper","rtpproxy_sock", "udp:rtp_proxy_publicIP:7890") 
modparam("nathelper","natping_interval", 30) 
modparam("nathelper","ping_nated_only", 0) 
modparam("nathelper","sipping_bflag", 7) 
modparam("nathelper","sipping_from", "sip:pinger at PROXY_IP" ) 
modparam("registrar","received_avp", "$avp(i:42)") 
modparam("nathelper","received_avp", "$avp(i:42)") 

I am running my RTP Proxy using the following command, 

./rtpproxy -l rtp_proxy_publicIP -s udp:*:7890 -F 

Thanks, 
Arjun 
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