[asterisk-users] Change Asterisk MulticastRTP codec

[Matthew Murphy]

Larry and Pete,
Thanks a bunch for jumping in and giving me some ideas! I am hoping to have something working soon with what you guys have given me. The end game for me is to be able to stream MP3s from a playlist. It appears like both solutions you guys have proposed may give me what I need. I will actually try both and let you know how it goes.
 --Matt

From: lmoore at omninet.net.au
To: asterisk-users at lists.digium.com
Date: Thu, 1 Oct 2015 06:15:17 +0800
Subject: Re: [asterisk-users] Change Asterisk MulticastRTP codec


  
    
  
  
    On my Asterisk 11 system I have the following in extensions.ael for
    chan_sip.

    

            8001    => {

                    Set(SIP_CODEC=alaw);

                   
    //Dial(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061);

                   
Page(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061,q,5);

                    Hangup();

            };

    

    

    I believe in your case you need to set PJSIP_MEDIA_OFFER(ulaw) in a
    pre-dial handler prior to making the call.

    

    See
    https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers+Specification.

    

    

    

    

    On 1/10/2015 1:51 AM, Matthew Murphy
      wrote:

    
    
      
      Greetings everyone,
        

        
        I was wondering if there was a way to change the codec that
          Asterisk uses when streaming via MulticastRTP. Or perhaps a
          way to transcode the multicast stream.
        

        
        In the CLI, when I have a multicast stream in progress, I
          am typing 'core show channel MulticastRTP/0x7f7........' to
          get lots of helpful information.
        

        
        I have noticed that when I do

          a MULTICAST page and send data from MP3Player, I get no
          sound on my speakers and get the following from 'core show
          channel PJSIP/xxx':
        

        
        
          NativeFormats: (slin)
          WriteFormat: slin
          ReadFormat: slin
          WriteTranscode: No 
          ReadTranscode: No 
        
        

        
        I have noticed that when I do a UNICAST page and send
            data from MP3Player, everything works flawlessly and I
          get the following from 'core show channel MulticastRTP':
        

        
        
          NativeFormats: (ulaw)
          WriteFormat: slin
          ReadFormat: slin
          WriteTranscode: Yes (slin at 8000)->(ulaw at 8000)
          ReadTranscode: Yes (ulaw at 8000)->(slin at 8000)
        
        

        
        

        
        The only thing that is changing is the following
          line in my extensions.conf file:
        

        
        ; For Multicast Paging
        
          same =>
            n(next),Page(MulticastRTP/basic/239.1.1.2:6061,q)
          

          
          ; For Unicast Paging
          same =>
n(next),Page(PJSIP/107&PJSIP/108,ib(pjsip-auto-answer-header^addheader^1)|p})
        
        

        
        

        
        Is there any way to get the MP3Player stream to transcode
          (as it does on the UNICAST stream) when I try to MULTICAST?
        

        
        Thanks for the help,
        

        
        --Matt
      
      

      
      

    
    

  


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