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&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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