Hello? I’m using Asterisk 20, at Rocky 9.4.
I’m trying to open Web/Mobile WebView Call service with JsSIP.
My Asterisk server has public ip. and apply ice/stun setting for mobile, web clients.
stun server setting is google’s thing. (stun.l.google.com:19302)
I have some unintelligible issue.
** I use only my cellphone’s network(5G). for my cellphone’s app, my laptop’s web(hotspot)…
- My laptop’s web → another user call [OK]
- another user → My laptop’s web [OK]
- My iPhone’s app(WebView) → another user call [ONE WAY RTP COMMUNICATE, cant hear sound both.]
- another user → My iPhone’s app(WebView) [OK]
- My iPhone’s web(chrome, safari…) → another user [ONE WAY RTP COMMUNICATE, cant hear sound both.]
- another user → My iPhone’s web(chrome, safari…) [OK]
I don’t know what is the problem.
I checked One way rtp communticate with ‘rtp set debug on’
packet (another user’s rtp → my iphone) is checked,
but packet (my iphone rtp → another user) isn’t checked.
But in OK situation, I checked two way rtp communicate for anyway platform.
Here is my asterisk pjsip endpoint settings.
And jssip code is just basic like docs.
localhost*CLI> pjsip show endpoint 1915
Endpoint: <Endpoint/CID…> <State…> <Channels.>
I/OAuth: <AuthId/UserName…>
Aor: <Aor…>
Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)..>
Transport: <TransportId…> <BindAddress…>
Identify: <Identify/Endpoint…>
Match: <criteria…>
Channel: <ChannelId…> <State…> <Time…>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>
Endpoint: 1915 Not in use 0 of inf
InAuth: 1915/1915
Aor: 1915 2
Contact: 1915/sip:u3emjibe@112.218.192.246:7 b323fa7fcf Avail 13.347
Transport: transport-wss wss 0 0 0.0.0.0:5063
ParameterName : ParameterValue
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (alaw|ulaw|g722|g729|opus)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : 1915
asymmetric_rtp_codec : false
auth : 1915
bind_rtp_to_media_address : false
bundle : true
call_group :
callerid :
callerid_privacy : allowed_not_screened
callerid_tag :
codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:pending, operation:union, keep:all, transcode:allow
connected_line_method : invite
contact_acl :
context : internal
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_auto_generate_cert : No
dtls_ca_file :
dtls_ca_path :
dtls_cert_file : /etc/asterisk/keys/fullchain.pem
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key : /etc/asterisk/keys/{Certificate of Authorized Authority}.key
dtls_rekey : 0
dtls_setup : actpass
dtls_verify : Yes
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
geoloc_incoming_call_profile :
geoloc_outgoing_call_profile :
ice_support : true
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language :
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : dtls
media_encryption_optimistic : false
media_use_received_transport : true
message_context :
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : no
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth :
outbound_proxy :
outgoing_call_offer_pref : remote_merge
overlap_context :
pickup_group :
preferred_codec_only : false
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : true
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
security_negotiation : no
send_aoc : false
send_connected_line : yes
send_diversion : true
send_history_info : false
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
stir_shaken : no
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
suppress_moh_on_sendonly : false
suppress_q850_reason_headers : false
t38_bind_udptl_to_media_address : false
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
tenantid :
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : transport-wss
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : true
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : yes
Please help me And I’m sorry because i can’t use english well.