I am getting the below error when I call from the web to web:
Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
Can someone help me to resolve this
here is the logs
582 [2021-04-14 11:35:03] ERROR[6108] res_pjsip_session.c: 2023: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
You would need to show the actual PJSIP configuration, as well as the SIP traces using “pjsip set logger on”. Based on the error it’s likely configuration.
Thanks for the quick response.
I am sharing configuration and pjsip logger
pjsip.endpoint.conf
[4000]
type=endpoint
aors=4000
auth=4000-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g729
context=from-internal
callerid=Mark2 <4000>
dtmf_mode=rfc4733
direct_media=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=yes
ice_support=no
media_use_received_transport=yes
trust_id_inbound=yes
user_eq_phone=yes
send_connected_line=yes
media_encryption=no
timers=yes
timers_min_se=90
media_encryption_optimistic=yes
refer_blind_progress=yes
refer_blind_progress=yes
rtp_timeout=20
rtp_timeout_hold=20
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
pjsip.transports.conf
[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=66.55.11.62
external_signaling_address=66.55.11.62
allow_reload=no
tos=cs3
cos=3
local_net=66.55.11.0/26
[0.0.0.0-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5060
external_media_address=66.55.11.62
external_signaling_address=66.55.11.62
allow_reload=no
tos=cs3
cos=3
local_net=66.55.11.0/26
[0.0.0.0-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
external_media_address=66.55.11.62
external_signaling_address=66.55.11.62
ca_list_file=/etc/pki/tls/certs/ca-bundle.crt
cert_file=/etc/asterisk/keys/default.pem
priv_key_file=/etc/asterisk/keys/default.key
method=default
verify_client=no
verify_server=no
allow_reload=no
tos=cs3
cos=3
local_net=66.55.11.0/26
[0.0.0.0-ws]
type=transport
protocol=ws
bind=0.0.0.0
external_media_address=66.55.11.62
external_signaling_address=66.55.11.62
allow_reload=no
tos=cs3
cos=3
local_net=66.55.11.0/26
[0.0.0.0-wss]
type=transport
protocol=wss
bind=0.0.0.0
external_media_address=66.55.11.62
external_signaling_address=66.55.11.62
allow_reload=no
tos=cs3
cos=3
local_net=66.55.11.0/26
//////////////////
gettting error
[2021-04-14 13:45:09] ERROR[6108]: res_pjsip_session.c:934 handle_incoming_sdp: 2023: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
<— Transmitting SIP response (350 bytes) to WS:182.73.26.18:60954 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS 192.0.2.171;rport=60954;received=182.73.26.18;branch=z9hG4bK4521147
Call-ID: hdnqe1mcc4ojg629t9ra
From: “Test Name” sip:2023@66.55.11.62;tag=cf87ub97bb
To: sip:4000@66.55.11.62;tag=8a7ea769-f354-46b5-9b2b-03f9aae45f97
CSeq: 7372 INVITE
Server: FPBX-15.0.17.24(16.15.1)
Content-Length: 0
Please check and what I am doing wrong in the configuration setting.
Please suggest .
That configuration is not correct for a WebRTC client. The wiki has an example[1]. As you are using, FreePBX, however, it’s up to you to figure out how to configure FreePBX as needed.
[1] Configuring Asterisk for WebRTC Clients - Asterisk Project - Asterisk Project Wiki
Thanks for your response.
can you please suggest what I am doing wrong in this configuration?
That particular error tends to be either because your media encryption settings are incompatible, or your allowed codecs are incompatible. However you seem to differ so much from the recommendation, that there may be other problem as well.
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