Hello there !
I’m trying to use WEBRTC with SIPJS and Asterisk (Asterisk GIT-16-89cf7899be) . It works very well with Chrome version ~50. Since I’m using chrome 96 on Android 10 audio seems not working at all…
Maybe codecs change?
This is my PJSIP configuration:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
[transport-ws]
type=transport
protocol=ws
bind=0.0.0.0
[1000]
type=aor
max_contacts=20
default_expiration=90400
minimum_expiration=86400
maximum_expiration=100000
[auth1000]
type=auth
auth_type=userpass
username=1000
password=1000
[1000]
type=endpoint
aors=1000
auth=auth1000
webrtc=yes
context=default
disallow=all
allow=ulaw
[998]
type=endpoint
context=default
allow=opus,alaw,ulaw,gsm,vp8,h264
dtls_auto_generate_cert=yes
webrtc=yes
auth=auth998
aors=998
[auth998]
type=auth
auth_type=userpass
password=998
username=998
[998]
type=aor
max_contacts=10
default_expiration=90400
minimum_expiration=86400
maximum_expiration=100000
[999]
type=endpoint
context=default
allow=opus,alaw,ulaw,gsm,vp8,h264
dtls_auto_generate_cert=yes
webrtc=yes
auth=auth999
aors=999
[auth999]
type=auth
auth_type=userpass
password=999
username=999
[999]
type=aor
max_contacts=10
default_expiration=90400
minimum_expiration=86400
maximum_expiration=100000
my RTP configuration
[general]
rtpstart=10000
rtpend=20000
My codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME FORMAT DESCRIPTION
------------------------------------------------------------------------------------------------
31 image png png (PNG Image)
6 audio g726 g726 (G.726 RFC3551)
4 audio alaw alaw (G.711 a-law)
2 audio g723 g723 (G.723.1)
20 audio speex speex (SpeeX)
21 audio speex speex16 (SpeeX 16khz)
22 audio speex speex32 (SpeeX 32khz)
24 audio g722 g722 (G722)
25 audio siren7 siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
32 video h261 h261 (H.261 video)
33 video h263 h263 (H.263 video)
8 audio adpcm adpcm (Dialogic ADPCM)
36 video h265 h265 (H.265 video)
44 audio silk silk8 (SILK Codec (8 KHz))
45 audio silk silk12 (SILK Codec (12 KHz))
46 audio silk silk16 (SILK Codec (16 KHz))
47 audio silk silk24 (SILK Codec (24 KHz))
28 audio g719 g719 (ITU G.719)
34 video h263p h263p (H.263+ video)
35 video h264 h264 (H.264 video)
19 audio g729 g729 (G.729A)
9 audio slin slin (16 bit Signed Linear PCM)
10 audio slin slin12 (16 bit Signed Linear PCM (12kHz))
11 audio slin slin16 (16 bit Signed Linear PCM (16kHz))
12 audio slin slin24 (16 bit Signed Linear PCM (24kHz))
13 audio slin slin32 (16 bit Signed Linear PCM (32kHz))
14 audio slin slin44 (16 bit Signed Linear PCM (44kHz))
15 audio slin slin48 (16 bit Signed Linear PCM (48kHz))
16 audio slin slin96 (16 bit Signed Linear PCM (96kHz))
17 audio slin slin192 (16 bit Signed Linear PCM (192kHz))
3 audio ulaw ulaw (G.711 u-law)
18 audio lpc10 lpc10 (LPC10)
27 audio testlaw testlaw (G.711 test-law)
43 audio none none (<Null> codec)
42 image t38 t38 (T.38 UDPTL Fax)
39 video vp9 vp9 (VP9 video)
38 video vp8 vp8 (VP8 video)
5 audio gsm gsm (GSM)
37 video mpeg4 mpeg4 (MPEG4 video)
23 audio ilbc ilbc (iLBC)
40 text red red (T.140 Realtime Text with redundancy)
41 text t140 t140 (Passthrough T.140 Realtime Text)
29 audio opus opus (Opus Codec)
30 image jpeg jpeg (JPEG image)
7 audio g726aal2 g726aal2 (G.726 AAL2)
1 audio codec2 codec2 (Codec 2)
26 audio siren14 siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
This is Asterisk console log output during a call (998 → 999):
Executing [999@default:1] NoOp("PJSIP/998-0000000a", "999") in new stack
Executing [999@default:2] Dial("PJSIP/998-0000000a", "PJSIP/999,60") in new stack
-- Called PJSIP/999
PJSIP/999-0000000b is ringing
PJSIP/999-0000000b is ringing
PJSIP/999-0000000b answered PJSIP/998-0000000a
> 0x7f9860016b70 -- Strict RTP learning after remote address set to: 194.xxx.xxx.xxx:44383
> 0x7f9860041470 -- Strict RTP learning after remote address set to: 213.xxx.xxx.xxx:58960
Channel PJSIP/999-0000000b joined 'simple_bridge' basic-bridge <2ef1f491-49dd-4fc8-b06f-6a01760baebb>
-- Channel PJSIP/998-0000000a joined 'simple_bridge' basic-bridge <2ef1f491-49dd-4fc8-b06f-6a01760baebb>
> 0x7f9860016b70 -- Strict RTP learning after ICE completion
> 0x7f9860041470 -- Strict RTP learning after ICE completion
-- Channel PJSIP/999-0000000b left 'simple_bridge' basic-bridge <2ef1f491-49dd-4fc8-b06f-6a01760baebb>
-- Channel PJSIP/998-0000000a left
I’m not able to see any problem from SIPJS (javascript console). There is just no soud 
Can someone help me with this issue ?