Hello I need help with my WEBRTC configuration,
when a webrtc client connects and makes a call
I get the following error :
[Dec 20 21:18:49] WARNING[201]: channel.c:5692 set_format: Unable to find a codec translation path: (g729) → (ulaw)
[Dec 20 21:18:49] WARNING[201]: channel.c:5692 set_format: Unable to find a codec translation path: (ulaw) → (g729)
– PJSIP/myitsp-0000001f is making progress passing it to PJSIP/105-0000001e
> 0x7f369806d350 – Strict RTP learning after remote address set to: 154.125.52.46:49667
> 0x7f3698054290 – Strict RTP switching to RTP target address 102.164.129.251:42486 as source
[Dec 20 21:18:49] WARNING[359][C-00000010]: translate.c:494 ast_translator_build_path: No translator path: (ending codec is not valid)
[Dec 20 21:18:49] WARNING[359][C-00000010]: translate.c:494 ast_translator_build_path: No translator path: (ending codec is not valid)
[Dec 20 21:18:49] WARNING[359][C-00000010]: channel.c:5692 set_format: Unable to find a codec translation path: (g729) → (opus)
my extensions.conf file is as follows:
[sortant]
exten=>7001,1,Dial(PJSIP/7001)
same=>n,Hangup()
exten=>7002,1,Dial(PJSIP/7002)
same=>n,Hangup()
exten=>105,1,Dial(PJSIP/105)
same=>n,Hangup()
exten=>106,1,Dial(PJSIP/105)
same=>n,Hangup()
exten =>_XXXXXXXXX,1,Wait(1) ;j’attends 1 seconde avant de creer un callid
same =>n,Set(CALLERID(all)= <+221328244040>)
same =>n,Dial(PJSIP/+221${EXTEN}@myitsp)
same =>n,Hangup()
And my pjsip.conf file is as follows :
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:15633
;Creation des utilisateurs WEB_RTC**
[105]
type=endpoint
aors=105
auth=105
use_avpf=yes
media_encryption=dtls
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=sortant
disallow=all
allow=opus,ulaw,vp9,vp8,h264
[105]
type = aor
max_contacts =25
[105]
type = auth
username = 105
password = 123