I’m looking to use the GSC3510 (latest firmware version) as part of a conferencing system on a LAN using asterisk and the opus codec. A GSC3510 will be placed on a table where a meeting will take place, and there will be people listening in another room with headphones via another enable sip phone.
I’m finding at the moment, even using the Opus codec, the audio quality from the microphone is low. If I make a recording on the local phone, it sounds fine, its just the audio over SIP/RTP seems low quality.
Because this is on a dedicated LAN, I’m not worried about it being high bandwidth.
Here is the outgoing sip/sdp from the phone.
v=0 o=5001 8000 8000 IN IP4 192.168.1.105 s=SIP Call c=IN IP4 192.168.1.105 t=0 0 m=audio 50040 RTP/AVP 107 101 a=sendrecv a=rtcp:50041 IN IP4 192.168.1.105 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
I’ve noticed the
sprop-maxcapturerate is set to
16000 can this be improved? Of course, there could be a problem somewhere else in the chain, but I thought I would start here.
Any help you can give to debug this issue would be appreciated. If this works out, we will be buying 10 of these units.