Hello,
I am looking to use several GSC3505 to make local radio system in a kindergarten (music + voice from the audio mixer). GSC3505 is convenient, because it has WiFi, so there is no need to arrange additional wiring for it.
I tried RTP multicast with G722 codec, but sound quality is very bad for music. RTSP music player plays MP3, but it is unreliable, e.g. it does not restart after nework failure.
The working solution is to make a call via baresip command line using opus codec. It seems to be convenient and reliable, but sound quality may have been much better.
As far as I understand, the problem is in maxcapturerate=16000 announced by GSC3505. Is there any way to override it either on the speaker itself, or in asterisk server? This would help a lot.
Thanks.
<--- SIP read from UDP:192.168.33.244:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.237:4970;branch=z9hG4bK1103c66c
From: <sip:0101@192.168.33.237:4970>;tag=as5b02ca80
To: <sip:1001@192.168.33.244:5060>;tag=458371582
Call-ID: 470fa39f2a6619d667c3c2f95fffaee1@192.168.33.237:4970
CSeq: 102 INVITE
Contact: <sip:1001@192.168.33.244:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream v 1.0.1.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 355
v=0
o=1001 8000 8000 IN IP4 192.168.33.244
s=SIP Call
c=IN IP4 192.168.33.244
t=0 0
m=audio 50040 RTP/AVP 107 101
a=sendrecv
a=rtcp:50041 IN IP4 192.168.33.244
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=0; sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0
a=maxptime:20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->