VoIP Mailing List Archives

[mrmdev at outlook.com]

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&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

[pete at fiberphone.co.nz]

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 <mrmdev@outlook.com (mrmdev@outlook.com)> wrote:

[lmoore at omninet.net.au]

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:

[mrmdev at outlook.com]

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@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: