I was working on asterisk 21 and come up with a plan to build webrtc based phone(web phone) and i tried it and works good. i was using zoiper as a secondary agent sip phone. and when i can from the zoiper to the web phone it works and i can hear audio in both side. but when i call from the webphone to the zoiper the connection will be established but the audio is one way, i can not hear a voice coming from the zoiper side. but i can hear the audio coming from the web phone in zoiper. i have tried my best to find the solution but i can’t. there are some other configuration like stun and NAT related thing to do but before that i wanted to see if there is a solution for this please. am at ubuntu 24
A solution requires understanding what is going on. First start by simplifying your scenario. Does calling from the webphone to Asterisk that just does a Playback work? If not, you’ve now narrowed it down. What is the SIP trace? What is the network arrangement? Is ICE negotiation and DTLS negotiation completing? Are RTP packets being sent from Asterisk to it?
WebRTC is built on various specifications and technology, and if you’re using/deploying it then most likely it will fail to work in some cases. An understanding of these is required to diagnose and understand what is going on.
Do you observe the same issue on the LAN? If yes , download another sip phone like portsip or microsip and test.if you observe the same issue, try calling Zoiper from portsip
Like the other contributor said ,check sip trace . It is likely to be codecs mismatch
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On Wed, Sep 24, 2025, 13:20 jcolp via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
jcolp Asterisk Project Lead
September 24A solution requires understanding what is going on. First start by simplifying your scenario. Does calling from the webphone to Asterisk that just does a Playback work? If not, you’ve now narrowed it down. What is the SIP trace? What is the network arrangement? Is ICE negotiation and DTLS negotiation completing? Are RTP packets being sent from Asterisk to it?
WebRTC is built on various specifications and technology, and if you’re using/deploying it then most likely it will fail to work in some cases. An understanding of these is required to diagnose and understand what is going on.
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here is what I did, my laptops window 11 and installed ubuntu server using virtual box and installed asterisk and nginx (for the webphone). I have configured a certificate for asterisk and nginx so that the webrtc can work with chrome(certificate problems ignored). I build a simple phone page using react with ‘jssip’ package. and build it and deployed it in the nginx server. then can one with the same home wifi can access the web-phone so I brought another window 11 pc and I opened chrome((certificate problems ignored) and access the webrtc. and the I tried to call from 1001 to 1002 and vice versa. they ring and the call establishes when answered perfectly. but the audio is one way. for example if 1001 makes the call the 1002 will hear audio but not 1001, some goes for 1002 when it calls 1001. every issue related to certificate and codec have been resolved. even I tried calling from zoiper to the webrtc and i can hear audio in both side. but when I call from the webphone to the zoiper there is no audio in the webphone. this is how I configured the agents,
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.pem
priv_key_file=/etc/asterisk/keys/asterisk.pem
method=tlsv1_2
external_media_address=192.168.1.6
external_signaling_address=192.168.1.6
local_net=192.168.0.0/24
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
allow_reload=yes
[1001]
type=endpoint
context=internal
disallow=all
allow=opus,ulaw,alaw,gsm
auth=1001-auth
aors=1001
transport=transport-wss
dtls_auto_generate_cert=yes
media_encryption=dtls
dtls_verify=fingerprint
webrtc=yes
ice_support=no
rtcp_mux=yes
use_avpf=yes
use_ptime=yes
media_use_received_transport=yes
direct_media=no
force_rport=yes
rewrite_contact=yes
[1001-auth]
type=auth
auth_type=userpass
username=1001
password=1234
[1001]
type=aor
max_contacts=10
[1002]
type=endpoint
context=internal
disallow=all
allow=opus,ulaw,alaw,gsm
auth=1002-auth
aors=1002
transport=transport-wss
dtls_auto_generate_cert=yes
media_encryption=dtls
webrtc=yes
ice_support=no
dtls_verify=fingerprint
rtcp_mux=yes
use_avpf=yes
media_use_received_transport=yes
direct_media=no
force_rport=yes
rewrite_contact=yes
use_ptime=yes
[1002-auth]
type=auth
auth_type=userpass
username=1002
password=1234
[1002]
type=aor
max_contacts=100
[1003]
type=endpoint
context=internal
disallow=all
allow=alaw
allow=gsm
allow=ulaw
allow=opus
auth=1003-auth
aors=1003
transport=transport-tls
direct_media=no
;media_encryption=sdes
[1003-auth]
type=auth
auth_type=userpass
username=1003
password=1234
[1003]
type=aor
max_contacts=1
[1004]
type=endpoint
context=internal
disallow=all
allow=alaw
allow=gsm
allow=alaw
allow=opus
auth=1004-auth
aors=1004
transport=transport-tls
direct_media=no
[1004-auth]
type=auth
auth_type=userpass
username=1004
password=1234
[1004]
type=aor
max_contacts=1
and the dial plan in the extension.conf
[internal]
exten => 1001,1,Dial(PJSIP/1001,200)
exten => 1001,n,Hangup()
exten => 1002,1,Dial(PJSIP/1002,200)
exten => 1002,n,Hangup()
exten => 1003,1,Dial(PJSIP/1003,200)
exten => 1003,n,Hangup()
exten => 1004,1,Dial(PJSIP/1004,200)
exten =? 1004,n,Hangup()
Generally what am experiencing is a one way audio issue.
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