Re: [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: lmo...@omninet.net.au
To: asterisk-users@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@8000)->(ulaw@8000)
  ReadTranscode: Yes (ulaw@8000)->(slin@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/107/108,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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[asterisk-users] Change Asterisk MulticastRTP codec

[Matthew Murphy]

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: slinReadFormat: slinWriteTranscode: 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: slinReadFormat: slinWriteTranscode: Yes 
(slin@8000)->(ulaw@8000)ReadTranscode: Yes (ulaw@8000)->(slin@8000)

The only thing that is changing is the following line in my extensions.conf 
file:
; For Multicast Pagingsame => n(next),Page(MulticastRTP/basic/239.1.1.2:6061,q)
; For Unicast Pagingsame => 
n(next),Page(PJSIP/107/108,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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Re: [asterisk-users] Change Asterisk MulticastRTP codec

[Pete Mundy]

Hi Matt

Interesting problem! I'm hoping those with knowledge about the internal 
workings of the Page app and multicast will chime in, although it might pay to 
quote your version of Asterisk).

I don't know enough to answer the question itself, but if it were me I would be 
inclined to just work around it by doing something like piping mp3player 
through sox before sending the data on to asterisk.

I may be able to help you achieve that, so if that's good enough then please 
post more of the multicast page config from your extensions.conf.

Pete


On 1/10/2015, at 6:51 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.
> [SNIP]
> Is there any way to get the MP3Player stream to transcode (as it does on the 
> UNICAST stream) when I try to MULTICAST?
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Re: [asterisk-users] Change Asterisk MulticastRTP codec

[Larry Moore]

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@8000)->(ulaw@8000)*
*ReadTranscode: Yes (ulaw@8000)->(slin@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/107/108,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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