Hello folks,
for a customer of us we are trying to make asterisk and windows RTC
library work together, but without success.
We *need* to use gsm codec, so in the "peer" section we have
disallow=all
allow=gsm
the sip signaling is ok, and when sniffing we got this session description:
INVITE from windows RTC:
v=0.
o=- 0 0 IN IP4 172.31.9.130.
s=session.
c=IN IP4 172.31.9.130.
b=CT:1000.
t=0 0.
m=audio 4632 RTP/AVP 97 111 112 6 0 8 4 5 3 101.
k=base64:ftJemQZ2pTDV5gzzqxG6ps5Ol5qiOt2qbP9L9Or5JQg.
a=rtpmap:97 red/8000.
a=rtpmap:111 SIREN/16000.
a=fmtp:111 bitrate=16000.
a=rtpmap:112 G7221/16000.
a=fmtp:112 bitrate=24000.
a=rtpmap:6 DVI4/16000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:5 DVI4/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=encryption:optional.
a=direction:active.
OK from asterisk 1.6 PBX:
v=0.
o=PBX 1705093286 1705093286 IN IP4 172.31.9.251.
s=PBX.
c=IN IP4 172.31.9.251.
t=0 0.
m=audio 14962 RTP/AVP 3 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
so, the rtp session should be GSM.
But the audio does not work.
In asterisk logs I see 'Got Siren7 offer at 24000 bps but only 32000
bps supported'.
any hint? anyone with the same issue?
unfortunately GSM is mandatory for us (we could not use alaw/ulaw,
that seems working).
thanks so much
stefano
