I’ve looked through other similar topics here and nothing looks similar to problem I have.
So, I’m using pjsip and Asterisk 20.5.2 built by buildozer @ build-3-19-x86_64 on a x86_64 running Linux on 2023-12-22 17:37:15 UTC on alpine linux 3.19 inside docker container.
I have two clients connected in following way:
“Intercom” client uses udp for both sip and rtp
# docker exec hosp-asterisk rasterisk -x 'pjsip show endpoint intercom1'
Endpoint: intercom1 Not in use 0 of inf
OutAuth: intercom1/intercom1
InAuth: intercom1/intercom1
Aor: intercom1 1
Contact: intercom1/sip:intercom1@192.168.3.165:5060 2d9e929e55 NonQual nan
Transport: transport-udp udp 0 0 0.0.0.0:5060
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (speex|g729|gsm|g723|ulaw|alaw|h264|h263)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : intercom1
asymmetric_rtp_codec : false
auth : intercom1
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : <unknown>
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 : intercoms-callout
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 :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain : 192.168.3.253
from_user :
g726_non_standard : false
geoloc_incoming_call_profile :
geoloc_outgoing_call_profile :
ice_support : false
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 : no
media_encryption_optimistic : false
media_use_received_transport : false
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 : intercom1
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 : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
security_mechanisms :
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 : off
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
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
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : transport-udp
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
and “User” client that uses tls/dtsl:
docker exec hosp-asterisk rasterisk -x 'pjsip show endpoint vlocal10011'
Endpoint: vlocal10011 Not in use 0 of inf
InAuth: vlocal10011/vlocal10011
Aor: vlocal10011 1
Contact: vlocal10011/sip:vlocal10011@192.168.3.137: 7d8cbb28e0 NonQual nan
Transport: transport-tls tls 0 0 0.0.0.0:5061
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (speex|g729|alaw|ulaw|h263|h264)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : vlocal10011
asymmetric_rtp_codec : false
auth : vlocal10011
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : <unknown>
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 : outbound-op
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 : /etc/ssl/certs/
dtls_cert_file : /run/secrets/asterisktlscert
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key : /run/secrets/asterisktlskey
dtls_rekey : 0
dtls_setup : actpass
dtls_verify : Yes
dtmf_mode : auto
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain : nightsterisk.devc.acme.com
from_user :
g726_non_standard : false
geoloc_incoming_call_profile :
geoloc_outgoing_call_profile :
ice_support : false
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 : 192.168.3.253
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 : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
security_mechanisms :
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 : off
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
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
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : transport-tls
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
When I try to make call from “User” to “Intercom” sip seems to work just fine and bridge is created on asterisk, but then suddenly call drops with following messages in the logs:
[2024-01-31 10:27:16.340] VERBOSE[768][C-00000003] app_dial.c: Called PJSIP/intercom1
[2024-01-31 10:27:16.465] VERBOSE[768][C-00000003] app_dial.c: PJSIP/intercom1-00000005 is ringing
[2024-01-31 10:27:16.469] VERBOSE[768][C-00000003] app_dial.c: PJSIP/intercom1-00000005 answered PJSIP/vlocal10011-00000004
[2024-01-31 10:27:16.473] VERBOSE[769][C-00000003] bridge_channel.c: Channel PJSIP/intercom1-00000005 joined 'simple_bridge' basic-bridge <ee84ed42-5a13-4b45-b849-aa8cc3c67375>
[2024-01-31 10:27:16.474] VERBOSE[768][C-00000003] bridge_channel.c: Channel PJSIP/vlocal10011-00000004 joined 'simple_bridge' basic-bridge <ee84ed42-5a13-4b45-b849-aa8cc3c67375>
[2024-01-31 10:27:17.009] ERROR[768][C-00000003] res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0x7f0fe3253ee0' due to reason 'called a function you should not call', terminating
[2024-01-31 10:27:17.010] WARNING[768][C-00000003] res_rtp_asterisk.c: RTP Read error: Invalid argument. Hanging up.
[2024-01-31 10:27:17.011] VERBOSE[768][C-00000003] bridge_channel.c: Channel PJSIP/vlocal10011-00000004 left 'simple_bridge' basic-bridge <ee84ed42-5a13-4b45-b849-aa8cc3c67375>
[2024-01-31 10:27:17.012] VERBOSE[769][C-00000003] bridge_channel.c: Channel PJSIP/intercom1-00000005 left 'simple_bridge' basic-bridge <ee84ed42-5a13-4b45-b849-aa8cc3c67375>
Error is
DTLS failure occurred on RTP instance '0x7f0fe3253ee0' due to reason 'called a function you should not call', terminating
This error message most likely comes from openssl library here:
which is probably called from asterisk code here:
I’ve tried this setup on lower and lower Asterisk versions and it eventually worked on Asterisk 16.6.2 (Alpine 3.11 in container).
So is this a bug in Asterisk or something wrong with my setup?