[OpenSIPS-Users] can't force rtpproxy use a new IP in SDP

Matt Hamilton mistral9999 at hotmail.com
Mon Jan 23 02:16:38 CET 2012


Hi Răzvan,

As I said in my previous email I got the  revision 8639 on 1.7.1 and that issue is fixed. However, I ran into a scenario where one of my scripts stopped working (audio issues).

In this scenario, I have a multihomed opensips with rtpproxy server bridging between opensips and and asterisk server.


 uac--------------------- opensips/rtpproxy           asterisk 

                           65.30.40.50

                           192.168.100.76--------192.168.100.101




Very briefly, my cfg looks like:

route:  
if (dst_ip == 65.30.40.50)    # (SRC: 1.2.3.4)  call coming in from public uac
    rtpproxy_offer("IE");


onreply:  
if (dst_ip == 192.168.100.76)  # (SRC: 192.168.100.101) asterisk replying
    rtpproxy_answer("IE");  

This runs fine with the rtpproxy.c that came with 1.7.1, but not with the revised rtpproxy.c

rtpproxy_answer places 192.168.100.76 instead of 65.30.40.50 in SDP (c) when replying to 1.2.3.4

I'm new to all this, so probably I'm doing something wrong, but jut wanted to let you know just in case. (I converted to the old rtpproxy.c, and it's working now.)

Regards,
Matt



Date: Sun, 22 Jan 2012 20:01:34 +0200
From: razvancrainea at opensips.org
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] can't force rtpproxy use a new IP in SDP


  


    
  
  
    Hi, Matt!

    

    The fix was only reported by Ovidiu and I was the one that fixed it
    in OpenSIPS's revision 8616 on trunk and 8639 on 1.7.1. This doesn't
    have anything to do with Kamailio. Can you please check if you have
    these changes? Because otherwise I don't think it will work.

    

    Regards,

    Răzvan

    

    On 01/22/2012 07:49 PM, Matt Hamilton wrote:
    
      
      
        Hi Răzvan,

        

        I'm using Opensips 1.7.1. I have seen Ovidiu's changes, but I
        think they are on the Kamailio side. if I'm not mistaken,
        sources are a little different, so I didn't want to mess with it
        (for now).

        

        I don't think there are any changes in Opensips (not in the
        trunk, either). 

        

        > Both scenarios you presented can be implemented using the
        rtpproxy module.

        

        Can I still use rtpproxy (without the fix) to get the first
        scenario going? Second one is all set.

        

        Thanks a lot,

        Matt

        

        

        
          Date: Sun, 22 Jan 2012 19:25:04 +0200

          From: razvancrainea at opensips.org

          To: users at lists.opensips.org

          Subject: Re: [OpenSIPS-Users] can't force rtpproxy use a new
          IP in SDP

          

          
          
          Hi, Matt!

          

          What version of OpenSIPS are you using? There was a bug
          reported by Ovidiu and fixed in revision 8639.

          Both scenarios you presented can be implemented using the
          rtpproxy module.

          

          Regards,

          Răzvan 

          

          On 01/22/2012 07:15 PM, Matt Hamilton wrote:
          
            
             Hi,

              

              I can't seem to
                force a new IP to change the connection in SDP; e.g.,

              

              rtpproxy_offer("c",

                "1.2.3.4"); 

              

              IP address I
                enter is always ignored. Any thoughts how I can resolve
                this?

                

                

                Here is my setup:

              

              private   NAT    
                  public             NAT      private       

              phones
                ----|------ internet ------------|---
                opensips/rtpproxy-----asterisk cluster

              

              

              I would like to force rtpproxy use the NATted public ip
              instead of the private ip of the opensips server when
              sending messages to UACs. advertised_address doesn't help
              btw.

              

              

              I was able to get the following going using the bridge
              mode. Can bridge mode be somehow used for the above?

              

              

              private   NAT      
                public                                NAT    private
                      

              phones
                ----|------ internet ------
                opensips/rtpproxy------|---asterisk cluster

              

              

              

              Thanks,

              Matt

              

              

              

            
            

            
            

            _______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

          
          

          _______________________________________________
          Users mailing list
          Users at lists.opensips.org
          http://lists.opensips.org/cgi-bin/mailman/listinfo/users
      
      

      
      

      _______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

    
  


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20120122/8842e4d5/attachment-0001.htm>


More information about the Users mailing list