PJSIP,JSSIP,NAT(behind firewall) have no audio on both side

RizeKishimaro · December 2, 2024, 4:57am

i’m on my way to publish the call center app to public internet.But it turns out both caller and callee
can’t hear audio and don’t transmit audio my asterisk setup is behind NAT(fortigate firewall)

192.168.130.20(local sip gateway) → 172.250.230.160(local call center) → bo.callcenter.xxx.xx(43.242.135.215(public internet))-> agents will use this with their public ip address I’m using asterisk 18(GIT-18-fe4394ebfe)

pjsip.conf


[global]
stunaddr=stun.l.google.com:19302
[general]
externip=43.242.135.215       
localnet=192.168.130.0/255.255.255.0  
localnet=19.168.0.0/32
bindaddr=0.0.0.0             
rtpstart=10000              
rtpend=20000  
[provider-template]
qualify = yes

[testnope]
qualify = yes

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
--;

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
external_media_address = 43.242.135.215
external_signaling_address = 43.242.135.215
local_net = 192.168.0.0/24
local_net = 172.250.230.0/32
local_net = 192.168.130.0/24

[testnope]
type = aor
contact = sip:192.168.130.20
qualify_frequency = 10

[testnope]
type = identify
endpoint = testnope
match = 192.168.130.20

[testnope]
type = endpoint
context = main-inbound
dtmf_mode = rfc4733
disallow = all
allow = ulaw,alaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
timers = no
direct_media = no
aors = testnope

[agent01_TEST]
type = aor
max_contacts = 10

[agent01_TEST]
type = auth
username = agent01_TEST
password = agt01

[agent01_TEST]
type = endpoint
context = testnope-outbound
disallow = all
allow = ulaw,alaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
ice_support = yes
direct_media = no
use_avpf = yes
rtcp_mux=yes
auth = agent01_TEST
outbound_auth = agent01_TEST
media_encryption = dtls
dtls_verify = fingerprint
dtls_ca_file= /etc/asterisk/keys/cert1.pem
dtls_cert_file =/etc/asterisk/keys/fullchain1.pem
dtls_private_key = /etc/asterisk/keys/privkey1.pem
dtls_setup = actpass
aors = agent01_TEST

rtp.conf

;
[general]
; Define RTP port range
rtpstart=10000
rtpend=20000

; Configure external NAT address
stunaddr=stun.l.google.com:19302

PJSIP endpoint

 Endpoint:  testnope                                             Not in use    0 of inf
        Aor:  testnope                                           0
      Contact:  testnope/sip:192.168.130.20                a87cbc347d Avail         1.897
   Identify:  testnope/testnope
        Match: 192.168.130.20/32


 Endpoint:  agent01_TEST                                         Not in use    0 of inf
    OutAuth:  agent01_TEST/agent01_TEST
     InAuth:  agent01_TEST/agent01_TEST
        Aor:  agent01_TEST                                      10
      Contact:  agent01_TEST/sip:p3jldbfr@69.160.29.28:342 11961691f4 NonQual         nan

PJSIP settings

Global Settings:

 ParameterName                              : ParameterValue
 ===========================================================================
 all_codecs_on_empty_reinvite               : false
 allow_sending_180_after_183                : false
 contact_expiration_check_interval          : 30
 debug                                      : no
 default_from_user                          : asterisk
 default_outbound_endpoint                  : default_outbound_endpoint
 default_realm                              : asterisk
 default_voicemail_extension                : 
 disable_multi_domain                       : false
 endpoint_identifier_order                  : ip,username,anonymous
 ignore_uri_user_options                    : false
 keep_alive_interval                        : 90
 max_forwards                               : 70
 max_initial_qualify_time                   : 0
 mwi_disable_initial_unsolicited            : false
 mwi_tps_queue_high                         : 500
 mwi_tps_queue_low                          : -1
 norefersub                                 : yes
 regcontext                                 : 
 send_contact_status_on_update_registration : no
 taskprocessor_overload_trigger             : global
 unidentified_request_count                 : 5
 unidentified_request_period                : 5
 unidentified_request_prune_interval        : 30
 use_callerid_contact                       : no
 user_agent                                 : Asterisk PBX GIT-18-fe4394ebfe

System Settings:

 ParameterName               : ParameterValue
 ============================================
 accept_multiple_sdp_answers : false
 compact_headers             : false
 disable_rport               : false
 disable_tcp_switch          : true
 follow_early_media_fork     : true
 threadpool_auto_increment   : 5
 threadpool_idle_timeout     : 60
 threadpool_initial_size     : 0
 threadpool_max_size         : 50
 timer_b                     : 32000
 timer_t1                    : 500

rtp settings


General Settings:
----------------
  Port start:      10000
  Port end:        20000
  Checksums:       Yes
  DTMF Timeout:    1200
  Strict RTP:      Yes
  Probation:       4 frames
  Replay Protect:  Yes
  ICE support:     Yes
  STUN address:    74.125.250.129:19302

stun status

asteriskdemo*CLI> stun show status
Hostname                  Port  Period  Retries  Status  ExternAddr       ExternPort
stun.l.google.com         19302 30      3        OK      43.242.135.215   14614

rtp log

Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026911, ts 3252809008, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013014, ts 3252809008, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026912, ts 3252809168, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013015, ts 3252809168, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026913, ts 3252809328, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013016, ts 3252809328, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026914, ts 3252809488, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013017, ts 3252809488, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026915, ts 3252809648, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013018, ts 3252809648, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026916, ts 3252809808, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013019, ts 3252809808, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026917, ts 3252809968, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013020, ts 3252809968, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026918, ts 3252810128, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013021, ts 3252810128, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026919, ts 3252810288, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013022, ts 3252810288, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026920, ts 3252810448, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013023, ts 3252810448, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026921, ts 3252810608, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013024, ts 3252810608, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026922, ts 3252810768, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013025, ts 3252810768, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026923, ts 3252810928, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013026, ts 3252810928, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026924, ts 3252811088, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013027, ts 3252811088, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026925, ts 3252811248, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013028, ts 3252811248, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026926, ts 3252811408, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013029, ts 3252811408, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026927, ts 3252811568, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013030, ts 3252811568, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026928, ts 3252811728, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013031, ts 3252811728, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026929, ts 3252811888, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013032, ts 3252811888, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026930, ts 3252812048, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013033, ts 3252812048, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026931, ts 3252812208, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013034, ts 3252812208, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026932, ts 3252812368, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013035, ts 3252812368, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026933, ts 3252812528, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013036, ts 3252812528, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026934, ts 3252812688, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013037, ts 3252812688, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026935, ts 3252812848, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013038, ts 3252812848, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026936, ts 3252813008, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013039, ts 3252813008, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026937, ts 3252813168, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013040, ts 3252813168, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026938, ts 3252813328, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013041, ts 3252813328, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026939, ts 3252813488, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013042, ts 3252813488, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026940, ts 3252813648, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013043, ts 3252813648, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026941, ts 3252813808, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013044, ts 3252813808, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026942, ts 3252813968, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013045, ts 3252813968, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026943, ts 3252814128, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013046, ts 3252814128, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026944, ts 3252814288, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013047, ts 3252814288, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026945, ts 3252814448, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013048, ts 3252814448, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026946, ts 3252814608, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013049, ts 3252814608, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026947, ts 3252814768, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013050, ts 3252814768, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026948, ts 3252814928, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013051, ts 3252814928, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026949, ts 3252815088, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013052, ts 3252815088, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026950, ts 3252815248, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013053, ts 3252815248, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026951, ts 3252815408, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013054, ts 3252815408, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026952, ts 3252815568, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013055, ts 3252815568, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026953, ts 3252815728, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013056, ts 3252815728, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026954, ts 3252815888, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013057, ts 3252815888, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026955, ts 3252816048, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013058, ts 3252816048, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026956, ts 3252816208, len 000160)
Sent RTP packet to      10.171.236.147:56736 (type 00, seq 013059, ts 3252816208, len 000160)
Got  RTP packet from    192.168.130.20:11346 (type 00, seq 026957, ts 3252816368, len 000160)

RizeKishimaro · December 2, 2024, 4:58am

If i connect to the network that asterisk is deploy and make calls i can hear the audio here is the logs

Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019182, ts 1095659208, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017971, ts 1019450528, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014709, ts 1019450528, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017283, ts 1095659368, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019183, ts 1095659368, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017972, ts 1019450688, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014710, ts 1019450688, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017284, ts 1095659528, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019184, ts 1095659528, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017973, ts 1019450848, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014711, ts 1019450848, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017285, ts 1095659688, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019185, ts 1095659688, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017974, ts 1019451008, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014712, ts 1019451008, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017286, ts 1095659848, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019186, ts 1095659848, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017975, ts 1019451168, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014713, ts 1019451168, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017287, ts 1095660008, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019187, ts 1095660008, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017976, ts 1019451328, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014714, ts 1019451328, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017288, ts 1095660168, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019188, ts 1095660168, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017977, ts 1019451488, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014715, ts 1019451488, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017289, ts 1095660328, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019189, ts 1095660328, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017978, ts 1019451648, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014716, ts 1019451648, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017290, ts 1095660488, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019190, ts 1095660488, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017979, ts 1019451808, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014717, ts 1019451808, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017291, ts 1095660648, len 000160)
Sent RTP packet to      172.250.230.36:54659 (via ICE) (type 00, seq 019191, ts 1095660648, len 000170)
Got  RTP packet from    172.250.230.36:54659 (type 00, seq 017980, ts 1019451968, len 000170)
Sent RTP packet to      192.168.130.20:18244 (type 00, seq 014718, ts 1019451968, len 000160)
Got  RTP packet from    192.168.130.20:18244 (type 00, seq 017292, ts 1095660808, len 000160)

Here is SIP logs

asteriskdemo*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Transmitting SIP request (437 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPj65a0ec69-888b-4dc3-817f-0f1f23df9ece
From: <sip:testnope@172.250.230.160>;tag=191102e4-b817-4070-bde0-50a2fdfd5ae5
To: <sip:192.168.130.20>
Contact: <sip:testnope@172.250.230.160:5060>
Call-ID: c1be5a7e-cd6a-47d0-a921-e3feab3c1406
CSeq: 51823 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (556 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPj65a0ec69-888b-4dc3-817f-0f1f23df9ece;received=172.250.230.160;rport=5060
From: <sip:testnope@172.250.230.160>;tag=191102e4-b817-4070-bde0-50a2fdfd5ae5
To: <sip:192.168.130.20>;tag=as4eddb63e
Call-ID: c1be5a7e-cd6a-47d0-a921-e3feab3c1406
CSeq: 51823 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


<--- Received SIP request (2086 bytes) from WSS:69.160.29.28:34235 --->
INVITE sip:09420042927@43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS u9l9460a53ah.invalid;branch=z9hG4bK4508526
Max-Forwards: 69
To: <sip:09420042927@43.242.135.215>
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
Call-ID: utvecv8cgd8cpd8g5c74
CSeq: 3479 INVITE
Contact: <sip:p3jldbfr@u9l9460a53ah.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: NextGenCC
Content-Length: 1544

v=0
o=- 6731843052318088297 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 5f1bdde1-780a-4a85-bdad-06d98b7a4650
m=audio 42382 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 10.171.236.147
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1426314780 1 udp 2122260223 10.171.236.147 42382 typ host generation 0 network-id 1 network-cost 900
a=candidate:734803076 1 tcp 1518280447 10.171.236.147 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:G8K1
a=ice-pwd:03kDRW1jbD/ZGqe19UISuyZz
a=ice-options:trickle
a=fingerprint:sha-256 8B:42:6D:5F:01:C3:47:7D:97:53:E5:34:34:AA:4A:ED:29:01:73:35:EA:8C:80:49:29:FA:02:D0:BE:B8:06:F0
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:5f1bdde1-780a-4a85-bdad-06d98b7a4650 df678379-371a-499d-8e89-c406c1e29d08
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2811900719 cname:FVhme3/PX07pvMWz
a=ssrc:2811900719 msid:5f1bdde1-780a-4a85-bdad-06d98b7a4650 df678379-371a-499d-8e89-c406c1e29d08

<--- Transmitting SIP response (492 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK4508526
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>;tag=z9hG4bK4508526
CSeq: 3479 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1733114775/b3d50280fab75bc63ed0187e5d8b9049",opaque="04af95ef23208981",algorithm=MD5,qop="auth"
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP request (419 bytes) from WSS:69.160.29.28:34235 --->
ACK sip:09420042927@43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS u9l9460a53ah.invalid;branch=z9hG4bK4508526
Max-Forwards: 69
To: <sip:09420042927@43.242.135.215>;tag=z9hG4bK4508526
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
Call-ID: utvecv8cgd8cpd8g5c74
CSeq: 3479 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: NextGenCC
Content-Length: 0


<--- Received SIP request (2375 bytes) from WSS:69.160.29.28:34235 --->
INVITE sip:09420042927@43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS u9l9460a53ah.invalid;branch=z9hG4bK7156426
Max-Forwards: 69
To: <sip:09420042927@43.242.135.215>
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
Call-ID: utvecv8cgd8cpd8g5c74
CSeq: 3480 INVITE
Authorization: Digest algorithm=MD5, username="agent01_TEST", realm="asterisk", nonce="1733114775/b3d50280fab75bc63ed0187e5d8b9049", uri="sip:09420042927@43.242.135.215", response="bbfdf5ee58183cde83ad36bd3b9143ac", opaque="04af95ef23208981", qop=auth, cnonce="idecpnrm2qmf", nc=00000001
Contact: <sip:p3jldbfr@u9l9460a53ah.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: NextGenCC
Content-Length: 1544

v=0
o=- 6731843052318088297 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 5f1bdde1-780a-4a85-bdad-06d98b7a4650
m=audio 42382 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 10.171.236.147
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1426314780 1 udp 2122260223 10.171.236.147 42382 typ host generation 0 network-id 1 network-cost 900
a=candidate:734803076 1 tcp 1518280447 10.171.236.147 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:G8K1
a=ice-pwd:03kDRW1jbD/ZGqe19UISuyZz
a=ice-options:trickle
a=fingerprint:sha-256 8B:42:6D:5F:01:C3:47:7D:97:53:E5:34:34:AA:4A:ED:29:01:73:35:EA:8C:80:49:29:FA:02:D0:BE:B8:06:F0
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:5f1bdde1-780a-4a85-bdad-06d98b7a4650 df678379-371a-499d-8e89-c406c1e29d08
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2811900719 cname:FVhme3/PX07pvMWz
a=ssrc:2811900719 msid:5f1bdde1-780a-4a85-bdad-06d98b7a4650 df678379-371a-499d-8e89-c406c1e29d08

<--- Transmitting SIP response (321 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK7156426
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>
CSeq: 3480 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


    -- Executing [09420042927@testnope-outbound:1] Set("PJSIP/agent01_TEST-0000000a", "CALLERID(num)=012399009") in new stack
    -- Executing [09420042927@testnope-outbound:2] Progress("PJSIP/agent01_TEST-0000000a", "") in new stack
    -- Executing [09420042927@testnope-outbound:3] Dial("PJSIP/agent01_TEST-0000000a", "PJSIP/09420042927@testnope,,KkTr") in new stack
       > 0x759cf8034370 -- Strict RTP learning after remote address set to: 10.171.236.147:42382
<--- Transmitting SIP response (1361 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK7156426
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
CSeq: 3480 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Contact: <sip:172.250.230.160:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Content-Type: application/sdp
Content-Length:   788

v=0
o=- 6731843052318088297 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
m=audio 17536 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:5c37a02220860a8178fce72a661e6c4c
a=ice-pwd:5f70c17164359d0c1769a951715cd020
a=candidate:Hacfae6a0 1 UDP 2130706431 172.250.230.160 17536 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 17536 typ host
a=candidate:S2bf287d2 1 UDP 1694498815 43.242.135.210 17536 typ srflx raddr 172.250.230.160 rport 17536
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux

    -- Called PJSIP/09420042927@testnope
<--- Transmitting SIP response (1361 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK7156426
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
CSeq: 3480 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:172.250.230.160:8089;transport=ws>
Content-Type: application/sdp
Content-Length:   788

v=0
o=- 6731843052318088297 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
m=audio 17536 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:5c37a02220860a8178fce72a661e6c4c
a=ice-pwd:5f70c17164359d0c1769a951715cd020
a=candidate:Hacfae6a0 1 UDP 2130706431 172.250.230.160 17536 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 17536 typ host
a=candidate:S2bf287d2 1 UDP 1694498815 43.242.135.210 17536 typ srflx raddr 172.250.230.160 rport 17536
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP request (929 bytes) to UDP:192.168.130.20:5060 --->
INVITE sip:09420042927@192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPjce4c96e0-1676-4bb2-951b-7233689304db
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>
Contact: <sip:asterisk@172.250.230.160:5060>
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29928 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, replaces, norefersub, histinfo
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Type: application/sdp
Content-Length:   267

v=0
o=- 1586072614 1586072614 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
m=audio 14790 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (544 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjce4c96e0-1676-4bb2-951b-7233689304db;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29928 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09420042927@192.168.130.20:5060>
Content-Length: 0


<--- Received SIP response (560 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjce4c96e0-1676-4bb2-951b-7233689304db;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>;tag=as4f017f1f
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29928 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09420042927@192.168.130.20:5060>
Content-Length: 0


    -- PJSIP/testnope-0000000b is ringing
<--- Transmitting SIP response (1361 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK7156426
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
CSeq: 3480 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:172.250.230.160:8089;transport=ws>
Content-Type: application/sdp
Content-Length:   788

v=0
o=- 6731843052318088297 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
m=audio 17536 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:5c37a02220860a8178fce72a661e6c4c
a=ice-pwd:5f70c17164359d0c1769a951715cd020
a=candidate:Hacfae6a0 1 UDP 2130706431 172.250.230.160 17536 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 17536 typ host
a=candidate:S2bf287d2 1 UDP 1694498815 43.242.135.210 17536 typ srflx raddr 172.250.230.160 rport 17536
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP request (437 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPj65d38f9b-4bc4-441c-bf9c-764e1e18275e
From: <sip:testnope@172.250.230.160>;tag=c6f8b9e4-8b15-4083-be7a-ad2e09212e98
To: <sip:192.168.130.20>
Contact: <sip:testnope@172.250.230.160:5060>
Call-ID: d64a8fb0-d76a-49ba-af16-51dd968f90d5
CSeq: 41236 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (556 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPj65d38f9b-4bc4-441c-bf9c-764e1e18275e;received=172.250.230.160;rport=5060
From: <sip:testnope@172.250.230.160>;tag=c6f8b9e4-8b15-4083-be7a-ad2e09212e98
To: <sip:192.168.130.20>;tag=as4b96261a
Call-ID: d64a8fb0-d76a-49ba-af16-51dd968f90d5
CSeq: 41236 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


<--- Received SIP response (870 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjce4c96e0-1676-4bb2-951b-7233689304db;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>;tag=as4f017f1f
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29928 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09420042927@192.168.130.20:5060>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 91283137 91283137 IN IP4 192.168.130.20
s=CAIP SIP 2.0
c=IN IP4 192.168.130.20
t=0 0
m=audio 10324 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

       > 0x759cf80212f0 -- Strict RTP learning after remote address set to: 192.168.130.20:10324
    -- PJSIP/testnope-0000000b is making progress passing it to PJSIP/agent01_TEST-0000000a
       > 0x759cf80212f0 -- Strict RTP switching to RTP target address 192.168.130.20:10324 as source
<--- Received SIP response (856 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjce4c96e0-1676-4bb2-951b-7233689304db;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>;tag=as4f017f1f
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29928 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09420042927@192.168.130.20:5060>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 91283137 91283137 IN IP4 192.168.130.20
s=CAIP SIP 2.0
c=IN IP4 192.168.130.20
t=0 0
m=audio 10324 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (428 bytes) to UDP:192.168.130.20:5060 --->
ACK sip:09420042927@192.168.130.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPjdec470d7-ad25-495e-8fac-efcd3916f90c
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>;tag=as4f017f1f
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29928 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


    -- PJSIP/testnope-0000000b answered PJSIP/agent01_TEST-0000000a
<--- Transmitting SIP response (1446 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK7156426
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
CSeq: 3480 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:172.250.230.160:8089;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   788

v=0
o=- 6731843052318088297 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
m=audio 17536 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:5c37a02220860a8178fce72a661e6c4c
a=ice-pwd:5f70c17164359d0c1769a951715cd020
a=candidate:Hacfae6a0 1 UDP 2130706431 172.250.230.160 17536 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 17536 typ host
a=candidate:S2bf287d2 1 UDP 1694498815 43.242.135.210 17536 typ srflx raddr 172.250.230.160 rport 17536
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux

    -- Channel PJSIP/testnope-0000000b joined 'simple_bridge' basic-bridge <a2054177-92e0-4c26-860d-726d1773be4d>
    -- Channel PJSIP/agent01_TEST-0000000a joined 'simple_bridge' basic-bridge <a2054177-92e0-4c26-860d-726d1773be4d>
<--- Received SIP request (448 bytes) from WSS:69.160.29.28:34235 --->
ACK sip:172.250.230.160:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS u9l9460a53ah.invalid;branch=z9hG4bK9826722
Max-Forwards: 69
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
Call-ID: utvecv8cgd8cpd8g5c74
CSeq: 3480 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: NextGenCC
Content-Length: 0


       > 0x759cf8034370 -- Strict RTP learning after ICE completion
       > 0x759cf80212f0 -- Strict RTP learning complete - Locking on source address 192.168.130.20:10324
<--- Transmitting SIP request (437 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPj7f676bbd-9cfd-4ab3-862a-ebff07c923ce
From: <sip:testnope@172.250.230.160>;tag=1844b4ee-ff0e-4de9-bb1d-f4e8b019c3b4
To: <sip:192.168.130.20>
Contact: <sip:testnope@172.250.230.160:5060>
Call-ID: df33a62b-e623-4670-8fb8-8fbcca33d13b
CSeq: 59915 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (556 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPj7f676bbd-9cfd-4ab3-862a-ebff07c923ce;received=172.250.230.160;rport=5060
From: <sip:testnope@172.250.230.160>;tag=1844b4ee-ff0e-4de9-bb1d-f4e8b019c3b4
To: <sip:192.168.130.20>;tag=as453f48ac
Call-ID: df33a62b-e623-4670-8fb8-8fbcca33d13b
CSeq: 59915 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


<--- Received SIP request (448 bytes) from WSS:69.160.29.28:34235 --->
BYE sip:172.250.230.160:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS u9l9460a53ah.invalid;branch=z9hG4bK6399353
Max-Forwards: 69
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
Call-ID: utvecv8cgd8cpd8g5c74
CSeq: 3481 BYE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: NextGenCC
Content-Length: 0


<--- Transmitting SIP response (355 bytes) to WSS:69.160.29.28:34235 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS u9l9460a53ah.invalid;rport=34235;received=69.160.29.28;branch=z9hG4bK6399353
Call-ID: utvecv8cgd8cpd8g5c74
From: <sip:agent01_TEST@43.242.135.215>;tag=6chrc9phqr
To: <sip:09420042927@43.242.135.215>;tag=0d0dd9ed-21a0-49c6-b6c4-65044fc97fc2
CSeq: 3481 BYE
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


    -- Channel PJSIP/agent01_TEST-0000000a left 'simple_bridge' basic-bridge <a2054177-92e0-4c26-860d-726d1773be4d>
  == Spawn extension (testnope-outbound, 09420042927, 3) exited non-zero on 'PJSIP/agent01_TEST-0000000a'
    -- Channel PJSIP/testnope-0000000b left 'simple_bridge' basic-bridge <a2054177-92e0-4c26-860d-726d1773be4d>
<--- Transmitting SIP request (452 bytes) to UDP:192.168.130.20:5060 --->
BYE sip:09420042927@192.168.130.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPja3ee5b01-4980-4112-a7da-f8fa274d5ee5
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>;tag=as4f017f1f
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29929 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (504 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPja3ee5b01-4980-4112-a7da-f8fa274d5ee5;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=9987e159-9bf4-4507-8ec8-a44b68fcdf91
To: <sip:09420042927@192.168.130.20>;tag=as4f017f1f
Call-ID: 9adcd313-0433-4441-a144-b00e998bf632
CSeq: 29929 BYE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<--- Transmitting SIP request (437 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPj7846f278-43a3-4960-9b0d-a71d0a3c8a59
From: <sip:testnope@172.250.230.160>;tag=d99e601c-07d1-49a4-b23a-16ec38232b97
To: <sip:192.168.130.20>
Contact: <sip:testnope@172.250.230.160:5060>
Call-ID: 982fb69a-6f09-4ef8-9609-3f1e8af2acae
CSeq: 11947 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0

jcolp · December 2, 2024, 9:26am

You need to do a packet capture and examine the ICE negotiation to see what is actually going on, and if anything is coming from the remote side. If the firewall doesn’t have ports forwarded then it very well could be blocking it, which would result in failure.

RizeKishimaro · December 2, 2024, 9:50am

why is it replying to 43.242.135.210

is it root of the problem?

jcolp · December 2, 2024, 9:53am

The root of the problem is that ICE negotiation did not complete and find a suitable way to communicate. ICE is a fundamental aspect of WebRTC, alongside DTLS-SRTP. If you’re deploying WebRTC, I highly highly suggest learning those fundamentals because when WebRTC goes wrong/doesn’t work that is the starting point.

ICE works with each side sending a list of candidates (IP address + port) that they can be reached at. Each side then sends STUN packets to the other sides’ candidates, and this is used to determine which work. If nothing gets through, then no path, no audio, no media.

Which then brings the question: why did it fail?

And that’s what you have to figure out.

RizeKishimaro · December 2, 2024, 10:13am

Thank You, @jcolp .One thing i want to know is why is the
ice candidate of 43.242.135.215 is not included we’re using fortigate

is there something wrong with the configuration of asterisk or the fortigate?

Edit: the 43.242.135.210 is firewall gateway ip

Thank you.

a=candidate:Hacfae6a0 1 UDP 2130706431 172.250.230.160 13694 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 13694 typ host
a=candidate:S2bf287d2 1 UDP 1694498815 43.242.135.210 13694 typ srflx raddr 172.250.230.160 rport 13694

jcolp · December 2, 2024, 10:15am

That IP address and port were discovered using the STUN server, so that’s how the traffic got to the STUN server - using that IP address and port which would mean that’s what the Fortigate did for the traffic.

External address can also be configured in rtp.conf for ICE candidates[1].

[1] asterisk/configs/samples/rtp.conf.sample at master · asterisk/asterisk · GitHub

RizeKishimaro · December 2, 2024, 10:25am

Thank you i will give it a try.

RizeKishimaro · December 2, 2024, 10:33am

How can i translate if the network traffic from .210 then i want send the media address to 215 how can i achieve it?

Kindly Rize.

RizeKishimaro · December 7, 2024, 5:44am

i figured out why 210 is fortigate firewall the firewall don’t allow to outbound traffic of rtp port 1000-20000 i will try about it

RizeKishimaro · December 9, 2024, 1:20pm

Somehow i can’t figure it out why is always exposing to private ip?The firewall has configured to accept ports rtp 10000-20000 udp 1-20000 tcp 1-9000 somehow it only use the private 172.250.230.160

v=0

o=- 4442532876196452110 2 IN IP4 172.250.230.160

s=Asterisk

c=IN IP4 172.250.230.160

a=candidate:H2bf287d7 1 UDP 2130706431 43.242.135.215 19092 typ host

a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 19092 typ host

a=candidate:S2bf287d2 1 UDP 1694498815 43.242.135.210 19092 typ srflx raddr 43.242.135.215 rport 19092

a=candidate:R2bf287d7 1 UDP 16777215 43.242.135.215 15262 typ relay raddr 192.168.128.1 rport 35129

a=rtpmap:0 PCMU/8000

i look through all forms related with NAT and PJSIP but no result have made any help would be appreciate

Siperb · December 9, 2024, 2:29pm

The line o=- 4442532876196452110 2 IN IP4 172.250.230.160 is not really used for WebRTC. It’s assumed that the web browser does not internally “know” you IP address, it needs to gather the IP address with the ICE operation, where it reaches an external server like google’s STUN server, and is able to return a message that contains your IP address.

You can test this process here:

These candidate are listed in order of priority; the higher in the list they are, the more efficient they are

Your firewall appears to be creating two candidates, one is
43.242.135.210 as type srflx: and the second (lower) is 43.242.135.215 as type relay.

srflx: The candidate is a server reflexive candidate; the ip and port are a binding allocated by a NAT for an agent when it sent a packet through the NAT to a server. They can be learned by the STUN server and TURN server to represent the candidate’s peer anonymously.

relay: The candidate is a relay candidate, obtained from a TURN server. The relay candidate’s IP address is an address the TURN server uses to forward the media between the two peers.

Also of interest, is the raddr attribute of the srflx entry means that there is a related IP 43.242.135.215 at play, so deciphering this, im assuming the firewall is creating some soft of relay service for you… on IP 43.242.135.215, and then advertising itself as 43.242.135.210 with a port forward of 43.242.135.210 to 43.242.135.215

I guess that since you are not aware of this, it may be some sort of default feature or service.
I’m pretty sure this can be disabled. But i’m not the firewall expert. It’s also seems overly complicated and could cause call quality problems with high traffic.

(I used a Fortigate firewall many years ago that required a lot of setup to get voip to work correctly over it. In the out-the-box config it simply could not keep up with the packet switching required, as it tries to hard to be “protecting” and the outcome is poor call quality.)

RizeKishimaro · December 9, 2024, 2:46pm

The Firewall is Fortinet and they disabled sip-helper.Really appreciate your answer Siperb.Your answer is very helpful.

but the stun monitor says

res_stun_monitor.c:149 stun_monitor_request: Old external address/port 0.0.0.0:0 now seen as 43.242.135.210:39941.

is this something related firewall doing things to asterisk?

Siperb · December 9, 2024, 3:05pm

res_stun_monitor.c:149 stun_monitor_request: Old external address/port 0.0.0.0:0 now seen as 43.242.135.210:39941.

Its Asterisk letting you know that the STUN monitoring (that the server does) has gone from no particular IP to 43.242.135.210. This normally happens a few minutes after the asterisk box is restarted. It’s just letting you know that it’s determined its own IP address.

RizeKishimaro · December 9, 2024, 3:09pm

there is a lot going on.Means there is no host to trickle so it fall back to it’s own IP?
which configuration i need to fix?

Siperb · December 9, 2024, 3:39pm

If I remember it was SIP-ALG… but again… not the expert on Fortinet

RizeKishimaro · December 9, 2024, 3:41pm

Thank you!Really Thank you. I will try to disable it and try again.Have a nice day.

RizeKishimaro · December 10, 2024, 5:00am

We disabled the SIP-ALG and SIP helper but it doesn’t change anything same as no audio no voice in both side

v=0
o=- 7485811073729612358 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS a0fa0924-0b95-451e-b82b-d581f45c71a2
m=audio 51243 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 10.169.247.197
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:286501867 1 udp 2122260223 10.169.247.197 51243 typ host generation 0 network-id 1 network-cost 900
a=candidate:4021911423 1 tcp 1518280447 10.169.247.197 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:ALvD
a=ice-pwd:7khfctE1qj/WSDWOuinUxwIi
a=ice-options:trickle
a=fingerprint:sha-256 85:21:6E:80:DF:D3:B5:99:24:1D:7F:85:E7:53:09:88:18:92:F7:0C:77:7E:B9:F1:6F:B2:8F:74:80:36:26:40
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:a0fa0924-0b95-451e-b82b-d581f45c71a2 7b598aa0-8030-4458-836b-60eeb138dab6
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2268228489 cname:xhZG+xzPDbarkCfo
a=ssrc:2268228489 msid:a0fa0924-0b95-451e-b82b-d581f45c71a2 7b598aa0-8030-4458-836b-60eeb138dab6

<--- Transmitting SIP response (493 bytes) to WSS:69.160.26.120:32825 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;rport=32825;received=69.160.26.120;branch=z9hG4bK7327760
Call-ID: tnqfim6pgvsb3sk1394j
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
To: <sip:09426941688@43.242.135.215>;tag=z9hG4bK7327760
CSeq: 5140 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1733806657/8d48fd10d1d8fd6b904cefec1da5f4d4",opaque="0d72959a09858c4e",algorithm=MD5,qop="auth"
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP request (419 bytes) from WSS:69.160.26.120:32825 --->
ACK sip:09426941688@43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;branch=z9hG4bK7327760
Max-Forwards: 69
To: <sip:09426941688@43.242.135.215>;tag=z9hG4bK7327760
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
Call-ID: tnqfim6pgvsb3sk1394j
CSeq: 5140 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: NextGenCC
Content-Length: 0


<--- Received SIP request (2375 bytes) from WSS:69.160.26.120:32825 --->
INVITE sip:09426941688@43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;branch=z9hG4bK3117450
Max-Forwards: 69
To: <sip:09426941688@43.242.135.215>
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
Call-ID: tnqfim6pgvsb3sk1394j
CSeq: 5141 INVITE
Authorization: Digest algorithm=MD5, username="agent01_TEST", realm="asterisk", nonce="1733806657/8d48fd10d1d8fd6b904cefec1da5f4d4", uri="sip:09426941688@43.242.135.215", response="c6ba24f4963693a91ce51524b7dedde5", opaque="0d72959a09858c4e", qop=auth, cnonce="tvnlf4crfv5k", nc=00000001
Contact: <sip:jg8h0m8u@ppfi62ga1aqu.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: NextGenCC
Content-Length: 1544

v=0
o=- 7485811073729612358 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS a0fa0924-0b95-451e-b82b-d581f45c71a2
m=audio 51243 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 10.169.247.197
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:286501867 1 udp 2122260223 10.169.247.197 51243 typ host generation 0 network-id 1 network-cost 900
a=candidate:4021911423 1 tcp 1518280447 10.169.247.197 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:ALvD
a=ice-pwd:7khfctE1qj/WSDWOuinUxwIi
a=ice-options:trickle
a=fingerprint:sha-256 85:21:6E:80:DF:D3:B5:99:24:1D:7F:85:E7:53:09:88:18:92:F7:0C:77:7E:B9:F1:6F:B2:8F:74:80:36:26:40
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:a0fa0924-0b95-451e-b82b-d581f45c71a2 7b598aa0-8030-4458-836b-60eeb138dab6
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2268228489 cname:xhZG+xzPDbarkCfo
a=ssrc:2268228489 msid:a0fa0924-0b95-451e-b82b-d581f45c71a2 7b598aa0-8030-4458-836b-60eeb138dab6

<--- Transmitting SIP response (322 bytes) to WSS:69.160.26.120:32825 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;rport=32825;received=69.160.26.120;branch=z9hG4bK3117450
Call-ID: tnqfim6pgvsb3sk1394j
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
To: <sip:09426941688@43.242.135.215>
CSeq: 5141 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


    -- Executing [09426941688@testnope-outbound:1] Set("PJSIP/agent01_TEST-00000016", "CALLERID(num)=012399009") in new stack
    -- Executing [09426941688@testnope-outbound:2] Progress("PJSIP/agent01_TEST-00000016", "") in new stack
    -- Executing [09426941688@testnope-outbound:3] Dial("PJSIP/agent01_TEST-00000016", "PJSIP/09426941688@webrtc,,KkTr") in new stack
       > 0x7b2f140693a0 -- Strict RTP learning after remote address set to: 10.169.247.197:51243
<--- Transmitting SIP response (1475 bytes) to WSS:69.160.26.120:32825 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;rport=32825;received=69.160.26.120;branch=z9hG4bK3117450
Call-ID: tnqfim6pgvsb3sk1394j
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
To: <sip:09426941688@43.242.135.215>;tag=a007f5d7-cbbe-4546-a5a6-1d3b3fae069e
CSeq: 5141 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Contact: <sip:172.250.230.160:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Content-Type: application/sdp
Content-Length:   901

v=0
o=- 7485811073729612358 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 19168 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:336ebf0e5eb507a32580dfc643deeed8
a=ice-pwd:088ffbfc134977782dff1a624576cbfc
a=candidate:H2bf287d7 1 UDP 2130706431 43.242.135.215 19168 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 19168 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:349340510 cname:b5e07b72-8a49-4037-8176-02b5d97521af
a=msid:65ff0c02-2faa-4dc7-8a27-8b21784aafa0 7e4bd11c-28e1-4d32-a419-cc4ed04f1e1f
a=rtcp-fb:* transport-cc
a=mid:0

    -- Called PJSIP/09426941688@webrtc
<--- Transmitting SIP response (1475 bytes) to WSS:69.160.26.120:32825 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;rport=32825;received=69.160.26.120;branch=z9hG4bK3117450
Call-ID: tnqfim6pgvsb3sk1394j
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
To: <sip:09426941688@43.242.135.215>;tag=a007f5d7-cbbe-4546-a5a6-1d3b3fae069e
CSeq: 5141 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:172.250.230.160:8089;transport=ws>
Content-Type: application/sdp
Content-Length:   901

v=0
o=- 7485811073729612358 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 19168 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:336ebf0e5eb507a32580dfc643deeed8
a=ice-pwd:088ffbfc134977782dff1a624576cbfc
a=candidate:H2bf287d7 1 UDP 2130706431 43.242.135.215 19168 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 19168 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:349340510 cname:b5e07b72-8a49-4037-8176-02b5d97521af
a=msid:65ff0c02-2faa-4dc7-8a27-8b21784aafa0 7e4bd11c-28e1-4d32-a419-cc4ed04f1e1f
a=rtcp-fb:* transport-cc
a=mid:0

<--- Transmitting SIP request (929 bytes) to UDP:192.168.130.20:5060 --->
INVITE sip:09426941688@192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPjf6ad689f-92c0-419a-a69d-ddd9ebb73c2f
From: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
To: <sip:09426941688@192.168.130.20>
Contact: <sip:asterisk@172.250.230.160:5060>
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 32429 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, replaces, norefersub, histinfo
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Type: application/sdp
Content-Length:   267

v=0
o=- 1601810988 1601810988 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
m=audio 13786 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (544 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjf6ad689f-92c0-419a-a69d-ddd9ebb73c2f;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
To: <sip:09426941688@192.168.130.20>
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 32429 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09426941688@192.168.130.20:5060>
Content-Length: 0


<--- Received SIP response (560 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjf6ad689f-92c0-419a-a69d-ddd9ebb73c2f;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
To: <sip:09426941688@192.168.130.20>;tag=as11267cf1
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 32429 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09426941688@192.168.130.20:5060>
Content-Length: 0


    -- PJSIP/webrtc-00000017 is ringing
<--- Transmitting SIP response (1475 bytes) to WSS:69.160.26.120:32825 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;rport=32825;received=69.160.26.120;branch=z9hG4bK3117450
Call-ID: tnqfim6pgvsb3sk1394j
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
To: <sip:09426941688@43.242.135.215>;tag=a007f5d7-cbbe-4546-a5a6-1d3b3fae069e
CSeq: 5141 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:172.250.230.160:8089;transport=ws>
Content-Type: application/sdp
Content-Length:   901

v=0
o=- 7485811073729612358 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 19168 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:336ebf0e5eb507a32580dfc643deeed8
a=ice-pwd:088ffbfc134977782dff1a624576cbfc
a=candidate:H2bf287d7 1 UDP 2130706431 43.242.135.215 19168 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 19168 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:349340510 cname:b5e07b72-8a49-4037-8176-02b5d97521af
a=msid:65ff0c02-2faa-4dc7-8a27-8b21784aafa0 7e4bd11c-28e1-4d32-a419-cc4ed04f1e1f
a=rtcp-fb:* transport-cc
a=mid:0

<--- Received SIP response (872 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjf6ad689f-92c0-419a-a69d-ddd9ebb73c2f;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
To: <sip:09426941688@192.168.130.20>;tag=as11267cf1
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 32429 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09426941688@192.168.130.20:5060>
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 988022488 988022488 IN IP4 192.168.130.20
s=CAIP SIP 2.0
c=IN IP4 192.168.130.20
t=0 0
m=audio 18850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

       > 0x7b2f140645d0 -- Strict RTP learning after remote address set to: 192.168.130.20:18850
    -- PJSIP/webrtc-00000017 is making progress passing it to PJSIP/agent01_TEST-00000016
       > 0x7b2f140645d0 -- Strict RTP switching to RTP target address 192.168.130.20:18850 as source
<--- Received SIP request (575 bytes) from WSS:204.157.172.104:15993 --->
REGISTER sip:43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS 8mo0ka6kp827.invalid;branch=z9hG4bK5512084
Max-Forwards: 69
To: <sip:agent01_TEST@43.242.135.215>
From: <sip:agent01_TEST@43.242.135.215>;tag=pprsp7854q
Call-ID: 2rr5dcpkqq31fmmfosmm86
CSeq: 3 REGISTER
Contact: <sip:t65h13mr@8mo0ka6kp827.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:d21581f4-4980-4105-9a92-01a029aaa054>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: NextGenCC
Content-Length: 0


<--- Transmitting SIP response (497 bytes) to WSS:204.157.172.104:15993 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 8mo0ka6kp827.invalid;rport=15993;received=204.157.172.104;branch=z9hG4bK5512084
Call-ID: 2rr5dcpkqq31fmmfosmm86
From: <sip:agent01_TEST@43.242.135.215>;tag=pprsp7854q
To: <sip:agent01_TEST@43.242.135.215>;tag=z9hG4bK5512084
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1733806658/ab518c84be60ef5544882cc60c6ddf3a",opaque="68f4c93b1b4af43d",algorithm=MD5,qop="auth"
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP request (852 bytes) from WSS:204.157.172.104:15993 --->
REGISTER sip:43.242.135.215 SIP/2.0
Via: SIP/2.0/WSS 8mo0ka6kp827.invalid;branch=z9hG4bK3088105
Max-Forwards: 69
To: <sip:agent01_TEST@43.242.135.215>
From: <sip:agent01_TEST@43.242.135.215>;tag=pprsp7854q
Call-ID: 2rr5dcpkqq31fmmfosmm86
CSeq: 4 REGISTER
Authorization: Digest algorithm=MD5, username="agent01_TEST", realm="asterisk", nonce="1733806658/ab518c84be60ef5544882cc60c6ddf3a", uri="sip:43.242.135.215", response="9ef4a7e5837636a1dba5dbbc66ae5cb8", opaque="68f4c93b1b4af43d", qop=auth, cnonce="540an3v525vo", nc=00000001
Contact: <sip:t65h13mr@8mo0ka6kp827.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:d21581f4-4980-4105-9a92-01a029aaa054>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: NextGenCC
Content-Length: 0


<--- Transmitting SIP response (676 bytes) to WSS:204.157.172.104:15993 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 8mo0ka6kp827.invalid;rport=15993;received=204.157.172.104;branch=z9hG4bK3088105
Call-ID: 2rr5dcpkqq31fmmfosmm86
From: <sip:agent01_TEST@43.242.135.215>;tag=pprsp7854q
To: <sip:agent01_TEST@43.242.135.215>;tag=z9hG4bK3088105
CSeq: 4 REGISTER
Date: Tue, 10 Dec 2024 04:57:38 GMT
Contact: <sip:jg8h0m8u@ppfi62ga1aqu.invalid;transport=ws>;expires=182
Contact: <sip:jg8h0m8u@ppfi62ga1aqu.invalid;transport=ws>;expires=236
Contact: <sip:8hefi2m5@8mo0ka6kp827.invalid;transport=ws>;expires=402
Contact: <sip:t65h13mr@8mo0ka6kp827.invalid;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Transmitting SIP request (437 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPj2c50988e-dd0a-4836-8301-9df452431ce3
From: <sip:testnope@172.250.230.160>;tag=26874420-4cf9-4f41-b850-7a90b656f08a
To: <sip:192.168.130.20>
Contact: <sip:testnope@172.250.230.160:5060>
Call-ID: 8219034b-5729-4c1a-b8f8-23fbd8f927a8
CSeq: 48315 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (556 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPj2c50988e-dd0a-4836-8301-9df452431ce3;received=172.250.230.160;rport=5060
From: <sip:testnope@172.250.230.160>;tag=26874420-4cf9-4f41-b850-7a90b656f08a
To: <sip:192.168.130.20>;tag=as507f13f3
Call-ID: 8219034b-5729-4c1a-b8f8-23fbd8f927a8
CSeq: 48315 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


       > 0x7b2f140645d0 -- Strict RTP learning complete - Locking on source address 192.168.130.20:18850
       > 0x7b2f140693a0 -- Strict RTP learning after ICE completion
<--- Transmitting SIP request (433 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPje9a2af38-5fe5-4783-81d2-041b9e4f63e8
From: <sip:webrtc@172.250.230.160>;tag=7be83ea3-e5eb-4cfc-b69a-f1171e022da2
To: <sip:192.168.130.20>
Contact: <sip:webrtc@172.250.230.160:5060>
Call-ID: 0364f93e-78c3-487e-a1fd-7d7129d4f783
CSeq: 29997 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (554 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPje9a2af38-5fe5-4783-81d2-041b9e4f63e8;received=172.250.230.160;rport=5060
From: <sip:webrtc@172.250.230.160>;tag=7be83ea3-e5eb-4cfc-b69a-f1171e022da2
To: <sip:192.168.130.20>;tag=as2244c90b
Call-ID: 0364f93e-78c3-487e-a1fd-7d7129d4f783
CSeq: 29997 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


<--- Received SIP request (543 bytes) from UDP:192.168.130.20:5060 --->
OPTIONS sip:172.250.230.160 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.20:5060;branch=z9hG4bK6b24af3c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.130.20>;tag=as798b0192
To: <sip:172.250.230.160>
Contact: <sip:asterisk@192.168.130.20:5060>
Call-ID: 528e337e6262b6912bf3d5d722c28b08@192.168.130.20:5060
CSeq: 102 OPTIONS
User-Agent: CAIP SIP 2.0
Date: Tue, 10 Dec 2024 04:36:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<--- Transmitting SIP response (857 bytes) to UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.20:5060;rport=5060;received=192.168.130.20;branch=z9hG4bK6b24af3c
Call-ID: 528e337e6262b6912bf3d5d722c28b08@192.168.130.20:5060
From: "asterisk" <sip:asterisk@192.168.130.20>;tag=as798b0192
To: <sip:172.250.230.160>;tag=z9hG4bK6b24af3c
CSeq: 102 OPTIONS
Accept: application/sdp, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/dialog-info+xml, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Transmitting SIP request (437 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPjf0f9947e-caf9-46cb-a275-4b12414b9e4e
From: <sip:testnope@172.250.230.160>;tag=6cc52d1d-3ddc-4ee1-bfba-0cb502f22781
To: <sip:192.168.130.20>
Contact: <sip:testnope@172.250.230.160:5060>
Call-ID: 1ac2998c-a5cd-4ab9-ab97-5161000e7bf4
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (556 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjf0f9947e-caf9-46cb-a275-4b12414b9e4e;received=172.250.230.160;rport=5060
From: <sip:testnope@172.250.230.160>;tag=6cc52d1d-3ddc-4ee1-bfba-0cb502f22781
To: <sip:192.168.130.20>;tag=as781d8773
Call-ID: 1ac2998c-a5cd-4ab9-ab97-5161000e7bf4
CSeq: 65409 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


<--- Received SIP response (858 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjf6ad689f-92c0-419a-a69d-ddd9ebb73c2f;received=172.250.230.160;rport=5060
From: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
To: <sip:09426941688@192.168.130.20>;tag=as11267cf1
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 32429 INVITE
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:09426941688@192.168.130.20:5060>
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 988022488 988022488 IN IP4 192.168.130.20
s=CAIP SIP 2.0
c=IN IP4 192.168.130.20
t=0 0
m=audio 18850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (428 bytes) to UDP:192.168.130.20:5060 --->
ACK sip:09426941688@192.168.130.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPj7f905b11-2e5b-4503-b050-047767e2aa24
From: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
To: <sip:09426941688@192.168.130.20>;tag=as11267cf1
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 32429 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


    -- PJSIP/webrtc-00000017 answered PJSIP/agent01_TEST-00000016
<--- Transmitting SIP response (1560 bytes) to WSS:69.160.26.120:32825 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;rport=32825;received=69.160.26.120;branch=z9hG4bK3117450
Call-ID: tnqfim6pgvsb3sk1394j
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
To: <sip:09426941688@43.242.135.215>;tag=a007f5d7-cbbe-4546-a5a6-1d3b3fae069e
CSeq: 5141 INVITE
Server: Asterisk PBX GIT-18-fe4394ebfe
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:172.250.230.160:8089;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   901

v=0
o=- 7485811073729612358 4 IN IP4 172.250.230.160
s=Asterisk
c=IN IP4 172.250.230.160
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 19168 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 22:C9:42:E8:F8:02:B5:BC:2D:61:BD:7B:A7:D5:F1:F3:6A:8D:DF:3B:DE:F9:56:C5:F2:1A:2E:9A:F8:63:C0:9A
a=ice-ufrag:336ebf0e5eb507a32580dfc643deeed8
a=ice-pwd:088ffbfc134977782dff1a624576cbfc
a=candidate:H2bf287d7 1 UDP 2130706431 43.242.135.215 19168 typ host
a=candidate:Ha6e7b90a 1 UDP 2130706431 fe80::20c:29ff:fe5a:734b 19168 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:349340510 cname:b5e07b72-8a49-4037-8176-02b5d97521af
a=msid:65ff0c02-2faa-4dc7-8a27-8b21784aafa0 7e4bd11c-28e1-4d32-a419-cc4ed04f1e1f
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/webrtc-00000017 joined 'simple_bridge' basic-bridge <35f3c074-e6a8-4e93-b11d-5b4c04386639>
    -- Channel PJSIP/agent01_TEST-00000016 joined 'simple_bridge' basic-bridge <35f3c074-e6a8-4e93-b11d-5b4c04386639>
<--- Received SIP request (448 bytes) from WSS:69.160.26.120:32825 --->
ACK sip:172.250.230.160:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS ppfi62ga1aqu.invalid;branch=z9hG4bK2766598
Max-Forwards: 69
To: <sip:09426941688@43.242.135.215>;tag=a007f5d7-cbbe-4546-a5a6-1d3b3fae069e
From: <sip:agent01_TEST@43.242.135.215>;tag=s5g58gi065
Call-ID: tnqfim6pgvsb3sk1394j
CSeq: 5141 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: NextGenCC
Content-Length: 0


<--- Transmitting SIP request (433 bytes) to UDP:192.168.130.20:5060 --->
OPTIONS sip:192.168.130.20 SIP/2.0
Via: SIP/2.0/UDP 172.250.230.160:5060;rport;branch=z9hG4bKPjc2e165ed-7642-4ef9-8caf-30db871d63d0
From: <sip:webrtc@172.250.230.160>;tag=7dfb31a3-4b04-4ea3-9fdc-5291eb711f3f
To: <sip:192.168.130.20>
Contact: <sip:webrtc@172.250.230.160:5060>
Call-ID: 2645d703-101b-42a6-8a77-aaa398265a5f
CSeq: 36975 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0


<--- Received SIP response (554 bytes) from UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.250.230.160:5060;branch=z9hG4bKPjc2e165ed-7642-4ef9-8caf-30db871d63d0;received=172.250.230.160;rport=5060
From: <sip:webrtc@172.250.230.160>;tag=7dfb31a3-4b04-4ea3-9fdc-5291eb711f3f
To: <sip:192.168.130.20>;tag=as5dca0dfe
Call-ID: 2645d703-101b-42a6-8a77-aaa398265a5f
CSeq: 36975 OPTIONS
Server: CAIP SIP 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.130.20:5060>
Accept: application/sdp
Content-Length: 0


<--- Received SIP request (447 bytes) from UDP:192.168.130.20:5060 --->
BYE sip:asterisk@172.250.230.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.20:5060;branch=z9hG4bK1090affc;rport
Max-Forwards: 70
From: <sip:09426941688@192.168.130.20>;tag=as11267cf1
To: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
CSeq: 102 BYE
User-Agent: CAIP SIP 2.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<--- Transmitting SIP response (369 bytes) to UDP:192.168.130.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.20:5060;rport=5060;received=192.168.130.20;branch=z9hG4bK1090affc
Call-ID: 3f71c7b9-18db-4465-8771-0e95946499c7
From: <sip:09426941688@192.168.130.20>;tag=as11267cf1
To: <sip:012399009@172.250.230.160>;tag=f301f469-e33d-481b-ae6c-b7baa166e526
CSeq: 102 BYE
Server: Asterisk PBX GIT-18-fe4394ebfe
Content-Length:  0

Siperb · December 10, 2024, 9:44am

The WebRTC phone, “NextGenCC” is sending an INVITE with an ICE candidate host IP:

a=candidate:286501867 1 udp 2122260223 10.169.247.197 51243 typ host

So Asterisk will attempt to send media to this address.

In its response, Asterisk is providing the Firewall IP of:

a=candidate:H2bf287d7 1 UDP 2130706431 43.242.135.215 19168 typ host

So firstly, the WebRTC side… It seems unlikely that the IP of 10.169.247.197 is reachable to the Asterisk server. Make sure that each WebRTC client, passes the ICE gather test in the link above. If you webrtc client is not “on the same LAN network” there has to be a candidate with type srflx.

Then, Asterisk side, you have rtp_symmetric on, but… Asterisk will build a bridge for this call, and media will flow in on 43.242.135.215 as per the candidate. However, If no media is received from the other side, no packets are sent, so the firewall does not build the NAT connection. So this means you can either push some packets out from the asterisk box, or you must force a port forward for the UDP RTP port range, from the firewall to the Asterisk box. In the past I used rtp_keepalive=1, that allowed media to flow over a NAT connection in a similar case.

jcolp · December 10, 2024, 9:55am

There doesn’t precisely have to be a srflx candidate. The ICE negotiation can result in the discovery of a peer reflexive candidate, that is: an incoming STUN packet from a source that does not match a candidate. Additionally even without media being forced like using rtp_keepalive the STUN traffic will still be flowing causing the NAT mapping to be established and maintained.

This does rely on the Asterisk side being reachable though, and I have concerns in that regard with the network arrangement based on the past comments.

Has there been a packet capture done on the Asterisk side to see if it is getting ANY STUN traffic? If the answer is no on both sides, then ideally you would determine why it is not getting to Asterisk and resolve that. Ideally one side is actually reachable, and further ideally that is Asterisk because trusting the client side is never great.

Topic		Replies	Views
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PJSIP,JSSIP,NAT(behind firewall) have no audio on both side

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