[OpenSIPS-Users] Media Issue once the UA - connected

Wed Aug 15 03:20:33 EDT 2018

 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 <bogdan at 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 at 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 at 159.200.37.234:5060;transport=UDP>. To: <sip:124007 at 149.355.453.253;transport=UDP>. From: <sip:124010 at 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 <sanjeevt510 at 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 <bogdan at 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 at 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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