Hello there,
I have just tried our fist webrtc test with Asterisk 16.19.1 + JsSIP 3.8.0
The endpoints get registered well and it’s able to send calls without problems but without audio.
Transport
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
local_net=192.168.1.23/24
external_media_address=211.37.146.88
external_signaling_address=211.37.146.88
allow_reload=yes
symmetric_transport=no
Endpoint
bcs16*CLI> pjsip show endpoint 7000
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <criteria.........................>
Channel: <ChannelId......................................> <State.....> <Time.....>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
==========================================================================================
Endpoint: 7000 Not in use 0 of inf
InAuth: 7000/7000
Aor: 7000 1
Contact: 7000/sip:7000@110.12.31.212:52195;transpor 3b1ec81810 Avail 12.972
Transport: transport-wss wss 0 0 0.0.0.0:5060
ParameterName : ParameterValue
==========================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode : 7000
acl :
aggregate_mwi : true
allow : (alaw|ulaw|opus)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : 7000
asymmetric_rtp_codec : false
auth : 7000
bind_rtp_to_media_address : false
bundle : true
call_group : 1
callerid : <unknown>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : outbound
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 : /etc/asterisk/letsencrypt/cert.pem
dtls_ca_path :
dtls_cert_file : /etc/asterisk/letsencrypt/cert.pem
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key : /etc/asterisk/letsencrypt/privkey.pem
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
ice_support : true
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
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 :
pickup_group : 1
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
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 : false
sub_min_expiry : 0
subscribe_context :
suppress_q850_reason_headers : false
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
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
bcs16*CLI>
DialPlan
exten => _01040230022,1,Noop(-- ${EXTEN} --)
same => n,Answer
same => n,Playback(vm-intro)
same => n,Hangup
This is the Asterisk CLI:
== WebSocket connection from '110.12.31.212:55771' for protocol 'sip' accepted using version '13'
-- Added contact 'sip:7000@110.12.31.212:55771;transport=ws' to AOR '7000' with expiration of 600 seconds
== Endpoint 7000 is now Reachable
-- Contact 7000/sip:7000@110.12.31.212:55771;transport=ws is now Reachable. RTT: 9.814 msec
== DTLS ECDH initialized (automatic), faster PFS enabled
-- Executing [01040230022@outbound:1] NoOp("PJSIP/7000-00000004", "-- 01040230022 --") in new stack
-- Executing [01040230022@outbound:2] Answer("PJSIP/7000-00000004", "") in new stack
> 0x7f713800ed20 -- Strict RTP learning after remote address set to: 110.12.31.212:54335
-- Executing [01040230022@outbound:3] Playback("PJSIP/7000-00000004", "vm-intro") in new stack
-- <PJSIP/7000-00000004> Playing 'vm-intro.gsm' (language 'en')
-- Executing [01040230022@outbound:4] Hangup("PJSIP/7000-00000004", "") in new stack
== Spawn extension (outbound, 01040230022, 4) exited non-zero on 'PJSIP/7000-00000004'
-- Executing [h@outbound:1] NoOp("PJSIP/7000-00000004", "CHANNEL=PJSIP/7000-00000004") in new stack
-- Executing [h@outbound:2] NoOp("PJSIP/7000-00000004", "CALLER="7000" <7000>") in new stack
bcs16*CLI>
RTP debug
bcs16*CLI> rtp set debug on
RTP Packet Debugging Enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
-- Executing [01040230022@outbound:1] NoOp("PJSIP/7000-00000001", "-- 01040230022 --") in new stack
-- Executing [01040230022@outbound:2] Answer("PJSIP/7000-00000001", "") in new stack
> 0x7fca500360d0 -- Strict RTP learning after remote address set to: 110.12.31.212:54486
-- Executing [01040230022@outbound:3] Playback("PJSIP/7000-00000001", "vm-intro") in new stack
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003067, ts 000160, len 000160)
-- <PJSIP/7000-00000001> Playing 'vm-intro.gsm' (language 'en')
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003068, ts 000320, len 000160)
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003069, ts 000480, len 000160)
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003070, ts 000640, len 000160)
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003071, ts 000800, len 000160)
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003072, ts 000960, len 000160)
Sent RTP packet to 110.12.31.212:54486 (type 08, seq 003073, ts 001120, len 000160)
...
I checked RTP Packet on my laptop with wireshark. but i can’t see a rtp
