Pjsip endpoints register timeouts

Asterisk Asterisk SIP
1

Hi
I have to pjsip recently. The endpoint registrations from the softphones have been working so far but from today the registrations are getting timeout. I dont see any registration request on the CLI.
I tried using different softphones and internet connections but still same. Sometimes it registers perfectly and sometime timeout.
I have around 1500 Pjsip endpoints on my asterisk and all of them are facing this issue.
I am using Asterisk certified/16.3-cert1 and PJPROJECT version currently running against: 2.8

Any help would be highly appreciated

2

Did you verified iptables rules and any other firewall, also make sure you re trying to register to the correct bind port, and verify the bind port on the Asterisk box with the netstat command

3

Yes there is no iptables and asterisk uptime is more than 15 days now.
here is the global and end point config

;;;;;;;;;;;;;;;;;;;;;;;;; GLOBAL ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
[global]
type = global
user_agent = “Call Engine 2.1”

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
external_media_address = x.x.x.x
external_signaling_address = x.x.x.x
external_signaling_port = 5060
tos = cs3

;;;;;;;;;;;;;;;; Endpoint ;;;;;;;;;;;;;;;;;;
[31259152]
type=aor
max_contacts=5
qualify_frequency=60

[31259152]
type=auth
username=31259152
password=23d9asda-0

[31259152]
type=endpoint
context=from-user1
disallow=all
allow=alaw,ulaw,vp8,h264
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
auth=31259152
outbound_auth=31259152
aors=31259152

4

Your transport section looks OK, as it seems the only one you have configured, enable sip trace and verify if the REGISTER request are sent, also run thet nestat commnd to verify the port is listening fine, also try using other transport section with a different port and configure it on a test endpoint

5

Okay i tried the above actions and its still the same!!! Here is the netstat output.
A quick observation from today, when i checked the status of endpoint it showed as Avail while on my softphone it was still trying to register and eventually timed-out.

netstat -tulnp
Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name
tcp 0 0 127.0.0.1:587 0.0.0.0:* LISTEN 1248/sendmail: MTA:
tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 24212/asterisk
tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN 1138/sshd
tcp 0 0 127.0.0.1:25 0.0.0.0:* LISTEN 1248/sendmail: MTA:
tcp 0 0 0.0.0.0:5666 0.0.0.0:* LISTEN 31902/nrpe
tcp6 0 0 :::22 :::* LISTEN 1138/sshd
tcp6 0 0 :::5666 :::* LISTEN 31902/nrpe
udp 0 0 0.0.0.0:5060 0.0.0.0:* 24212/asterisk
udp 0 0 0.0.0.0:36462 0.0.0.0:* 24212/asterisk
udp 0 0 0.0.0.0:68 0.0.0.0:* 915/dhclient
udp6 0 0 :::46974 :::* 24212/asterisk

;;;;;;;;;;;;;;;;;;;;;;;;;
Endpoint: 68217599 Not in use 0 of inf
OutAuth: 68217599/68217599
InAuth: 68217599/68217599
Aor: 68217599 5
Contact: 68217599/sip:68217599@167.99.119.244:44029 5a8f477108 Avail 74.400

ParameterName : ParameterValue

100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (alaw|ulaw|vp8|h264)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 68217599
asymmetric_rtp_codec : false
auth : 68217599
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid :
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : from-user1
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 :
from_user :
g726_non_standard : false
ice_support : false
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 : 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 : 68217599
outbound_proxy :
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 : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_connected_line : yes
send_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : 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 :
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

6

hi does any one has the idea whats going wrong in here.
when i switch back to sip i dont face this problem.
this is my production traffic and I cant keep it like that.
Any help would be highly appreciated

7

When you try sip it is on the same bind port ? also have you try to use a test BOX with the latest version of Asterisk (LTS) and the PJSIP

8

Yes sip Bind port is same (5060).
I have now reinstalled the LTS version but the problem seems to remain the same.
One thing i may forgot to mention is the server is currently hosting around 2000 endpoints. Could this be causing registrations to timeout?

9

Are there any other warning or error messages? I’d expect some complaints about the thread pool…

Either analyze the log files, or start asterisk with “asterisk -c” to see only the most important messages.

10

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