Call dropping and one way audio

JohnDo · June 6, 2020, 3:50pm

Hello guys,

I’m a newbie to Asterisk and I need your help.

I installed Asterisk 17.0.1 in AWS on a AMI instance. I configured four users (usrermobil, userpc, outuser, mizudroid). With userpc I’m authenticated from Zoiper on my laptop, with outuser on MicroSIP on my laptop, with usermobil eith Zoiper on my phone and with mizudroid on Mizudriod app on my phone. The only call working with audio on both sides and not dropping is from the mizudroid user to any other. The settings in each softphone are as default, no STUN. My mobile phone is on the same wifi as the laptop.
I’m attaching a screenshot with the messages received in Astersik CLI when calling from zoiper app from the mobile, to zoiper on my laptop and the call is dropping after 7s and also there is one way audio from my laptop to the phone

.

SIP.conf

[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
qualify=yes

[usermobil]
type=friend
context=phones
allow=ulaw,alaw
secret=pass1234
host=dynamic

[userpc]
type=friend
context=phones
allow=ulaw,alaw
secret=pass1234
host=dynamic

[outuser]
type=friend
context=incoming
allow=ulaw,alaw
secret=pass1234
host=dynamic

[mizudroid]
type=friend
context=phones
allow=ulaw,alaw
secret=pass1234
host=dynamic

Extensions.conf
[incoming]

exten => 744500,1,Goto(phones,100,1)

[outbound]

exten => _X.,1,Dial(SIP/outuser)

[phones]

exten => 100,1,NoOp(Display a line)
same => n,Dial(SIP/userpc)
same => n,Hangup

exten => 200,1,NoOp(Display a line)
;same => n,Playback(tt-monkeys)
same => n,Dial(SIP/usermobil)
same => n,Hangup

exten => 300,1,NoOp(Display a line)
same => n,Dial(SIP/mizudroid)
same => n,Hangup

exten => _0X.,1,NoOp({{EXTEN}}) same => n,Goto(outbound,{EXTEN:1},1)

david551 · June 6, 2020, 4:31pm

Please post logs as text, so they can be searched and cut and pasted.

You have a fatal transmission timeout, which means that a packet is not reaching its desitnation in one direction or another.

The phone either failed to received the 200 OK, or received it and failed to send an ACK to the right place., or at all.

JohnDo · June 6, 2020, 7:29pm

It wont allow me to upload the txt as I’m new user and to copy all the lines is too long, over 32000 characters.

JohnDo · June 6, 2020, 7:42pm

This would be the first part


<--- SIP read from UDP:188.27.130.120:26817 --->
INVITE sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;rport
Max-Forwards: 70
Contact: <sip:usermobil@188.27.130.120:26817;transport=UDP>
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.4.4
Authorization: Digest username="usermobil",realm="asterisk",nonce="3da3eb2a",uri="sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP",response="2a2889c45779738744ab393ead6acbbd",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 259

v=0
o=Zoiper 1591466117822 1 IN IP4 192.168.0.102
s=Z
c=IN IP4 192.168.0.102
t=0 0
m=audio 49936 RTP/AVP 3 101 0 8 97 110
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:110 speex/8000
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 188.27.130.120:26817 (NAT)
Using INVITE request as basis request - k1-dUQgFfBN9Sl-QS4zu2A..
Found peer 'usermobil' for 'usermobil' from 188.27.130.120:26817
  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 110
Found audio description format telephone-event for ID 101
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.102:49936
Looking for 100 in phones (domain ec2-18-184-51-208.eu-central-1.compute.amazonaws.com)
sip_route_dump: route/path hop: <sip:usermobil@188.27.130.120:26817;transport=UDP>

<--- Transmitting (NAT) to 188.27.130.120:26817 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Length: 0


<------------>
    -- Executing [100@phones:1] NoOp("SIP/usermobil-0000005a", "Display a line") in new stack
    -- Executing [100@phones:2] Dial("SIP/usermobil-0000005a", "SIP/userpc") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 18418
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 188.27.130.120:27031:
INVITE sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917 SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK229451c4
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 17.1.0
Date: Sat, 06 Jun 2020 17:55:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 2021932168 2021932168 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 18418 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/userpc
Retransmitting #1 (no NAT) to 188.27.130.120:27031:
INVITE sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917 SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK229451c4
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 17.1.0
Date: Sat, 06 Jun 2020 17:55:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 2021932168 2021932168 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 18418 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK229451c4;received=18.184.51.208
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK229451c4;received=18.184.51.208
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK229451c4;received=18.184.51.208
Contact: <sip:userpc@188.27.130.120:27031;transport=UDP>
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.7 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:userpc@188.27.130.120:27031;transport=UDP>
    -- SIP/userpc-0000005b is ringing

<--- Transmitting (NAT) to 188.27.130.120:26817 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK229451c4;received=18.184.51.208
Contact: <sip:userpc@188.27.130.120:27031;transport=UDP>
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.7 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 424

v=0
o=Z 0 3 IN IP4 188.27.130.120
s=Z
c=IN IP4 188.27.130.120
t=0 0
m=audio 8000 RTP/AVP 0 8 106 9 3 97 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 188.27.130.120:8000
sip_route_dump: route/path hop: <sip:userpc@188.27.130.120:27031;transport=UDP>
set_destination: Parsing <sip:userpc@188.27.130.120:27031;transport=UDP> for address/port to send to
set_destination: set destination to 188.27.130.120:27031
Transmitting (no NAT) to 188.27.130.120:27031:
ACK sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK70625cc2
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 17.1.0
Content-Length: 0

JohnDo · June 6, 2020, 7:42pm

And the second


---
    -- SIP/userpc-0000005b answered SIP/usermobil-0000005a
Audio is at 55244
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 188.27.130.120:26817 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/userpc-0000005b joined 'simple_bridge' basic-bridge <ff09a82b-2f1f-435a-9f3d-4b40b1846621>
    -- Channel SIP/usermobil-0000005a joined 'simple_bridge' basic-bridge <ff09a82b-2f1f-435a-9f3d-4b40b1846621>
set_destination: Parsing <sip:userpc@188.27.130.120:27031;transport=UDP> for address/port to send to
set_destination: set destination to 188.27.130.120:27031
Audio is at 18418
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 188.27.130.120:27031:
INVITE sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK2db8797e
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 2021932168 2021932169 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 192.168.0.102
t=0 0
m=audio 49936 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
Retransmitting #1 (NAT) to 188.27.130.120:26817:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
Retransmitting #1 (no NAT) to 188.27.130.120:27031:
INVITE sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK2db8797e
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 2021932168 2021932169 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 192.168.0.102
t=0 0
m=audio 49936 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK2db8797e;received=18.184.51.208
Contact: <sip:userpc@188.27.130.120:27031;transport=UDP>
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.7 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 424

v=0
o=Z 0 6 IN IP4 188.27.130.120
s=Z
c=IN IP4 188.27.130.120
t=0 0
m=audio 8000 RTP/AVP 0 8 106 9 3 97 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 188.27.130.120:8000
set_destination: Parsing <sip:userpc@188.27.130.120:27031;transport=UDP> for address/port to send to
set_destination: set destination to 188.27.130.120:27031
Transmitting (no NAT) to 188.27.130.120:27031:
ACK sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK2d734591
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 17.1.0
Content-Length: 0


---
Retransmitting #2 (NAT) to 188.27.130.120:26817:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
Retransmitting #3 (NAT) to 188.27.130.120:26817:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
Really destroying SIP dialog 'mPVRrIkuD67tnEopJsfGwg..' Method: REGISTER
Retransmitting #4 (NAT) to 188.27.130.120:26817:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:188.27.130.120:27087 --->

<------------->
Reliably Transmitting (NAT) to 188.27.130.120:27087:
OPTIONS sip:outuser@192.168.0.103:53531;ob SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK3c191c35;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.31.44.215>;tag=as49fc466e
To: <sip:outuser@192.168.0.103:53531;ob>
Contact: <sip:asterisk@172.31.44.215:5060>
Call-ID: 057bc25d42332dd87242fb8c082a1860@172.31.44.215:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 17.1.0
Date: Sat, 06 Jun 2020 17:55:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:188.27.130.120:27087 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;rport=5060;received=18.184.51.208;branch=z9hG4bK3c191c35
Call-ID: 057bc25d42332dd87242fb8c082a1860@172.31.44.215:5060
From: "asterisk" <sip:asterisk@172.31.44.215>;tag=as49fc466e
To: <sip:outuser@192.168.0.103;ob>;tag=z9hG4bK3c191c35
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.26
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '057bc25d42332dd87242fb8c082a1860@172.31.44.215:5060' Method: OPTIONS
Retransmitting #5 (NAT) to 188.27.130.120:26817:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
Retransmitting #6 (NAT) to 188.27.130.120:26817:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---812b9a2a311cb2fd;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
To: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 2 INVITE
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:100@172.31.44.215:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965027 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
[Jun  6 17:55:31] WARNING[13182]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission k1-dUQgFfBN9Sl-QS4zu2A.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Jun  6 17:55:31] WARNING[13182]: chan_sip.c:4143 retrans_pkt: Hanging up call k1-dUQgFfBN9Sl-QS4zu2A.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/usermobil-0000005a left 'native_rtp' basic-bridge <ff09a82b-2f1f-435a-9f3d-4b40b1846621>
Audio is at 55244
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 188.27.130.120:26817:
INVITE sip:usermobil@188.27.130.120:26817;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK34e80214;rport
Max-Forwards: 70
From: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
Contact: <sip:100@172.31.44.215:5060>
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 102 INVITE
User-Agent: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965028 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
set_destination: Parsing <sip:userpc@188.27.130.120:27031;transport=UDP> for address/port to send to
set_destination: set destination to 188.27.130.120:27031
Audio is at 18418
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 188.27.130.120:27031:
INVITE sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK63dff13f
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 2021932168 2021932170 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 18418 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
  == Spawn extension (phones, 100, 2) exited non-zero on 'SIP/usermobil-0000005a'
Scheduling destruction of SIP dialog 'k1-dUQgFfBN9Sl-QS4zu2A..' in 6400 ms (Method: INVITE)
    -- Channel SIP/userpc-0000005b left 'native_rtp' basic-bridge <ff09a82b-2f1f-435a-9f3d-4b40b1846621>
Scheduling destruction of SIP dialog '43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:188.27.130.120:26817 --->


<------------->

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK63dff13f;received=18.184.51.208
Contact: <sip:userpc@188.27.130.120:27031;transport=UDP>
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 104 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.7 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 424

v=0
o=Z 0 9 IN IP4 188.27.130.120
s=Z
c=IN IP4 188.27.130.120
t=0 0
m=audio 8000 RTP/AVP 0 8 106 9 3 97 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 188.27.130.120:8000
set_destination: Parsing <sip:userpc@188.27.130.120:27031;transport=UDP> for address/port to send to
set_destination: set destination to 188.27.130.120:27031
Transmitting (no NAT) to 188.27.130.120:27031:
ACK sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK34cef21c
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Contact: <sip:usermobil@172.31.44.215:5060>
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 17.1.0
Content-Length: 0


---
set_destination: Parsing <sip:userpc@188.27.130.120:27031;transport=UDP> for address/port to send to
set_destination: set destination to 188.27.130.120:27031
Reliably Transmitting (no NAT) to 188.27.130.120:27031:
BYE sip:userpc@188.27.130.120:27031;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK0ed46db4
Max-Forwards: 70
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 105 BYE
User-Agent: Asterisk PBX 17.1.0
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
Scheduling destruction of SIP dialog '43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 188.27.130.120:26817:
INVITE sip:usermobil@188.27.130.120:26817;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK34e80214;rport
Max-Forwards: 70
From: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
Contact: <sip:100@172.31.44.215:5060>
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 102 INVITE
User-Agent: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 924965027 924965028 IN IP4 172.31.44.215
s=Asterisk PBX 17.1.0
c=IN IP4 172.31.44.215
t=0 0
m=audio 55244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:188.27.130.120:27031 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK0ed46db4;received=18.184.51.208
Contact: <sip:userpc@188.27.130.120:27031;transport=UDP>
To: <sip:userpc@188.27.130.120:27031;transport=UDP;rinstance=9f74a0f26f038917>;tag=33352613
From: <sip:usermobil@172.31.44.215>;tag=as7385395e
Call-ID: 43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060
CSeq: 105 BYE
User-Agent: Z 5.3.7 rv2.9.30
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '43f6c3f843be61db2b1311c733ef0dc7@172.31.44.215:5060' Method: INVITE

<--- SIP read from UDP:188.27.130.120:26817 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK34e80214;rport=5060;received=18.184.51.208
Contact: <sip:usermobil@188.27.130.120:26817;transport=UDP>
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
From: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.4.4
Allow-Events: presence, kpml, talk
Content-Length: 247

v=0
o=Zoiper 0 2 IN IP4 192.168.0.102
s=Z
c=IN IP4 192.168.0.102
t=0 0
m=audio 49936 RTP/AVP 0 3 8 97 110 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 101
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.102:49936
Transmitting (NAT) to 188.27.130.120:26817:
ACK sip:usermobil@188.27.130.120:26817;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK191ff45a;rport
Max-Forwards: 70
From: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
Contact: <sip:100@172.31.44.215:5060>
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 102 ACK
User-Agent: Asterisk PBX 17.1.0
Content-Length: 0


---
Reliably Transmitting (NAT) to 188.27.130.120:26817:
BYE sip:usermobil@188.27.130.120:26817;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK0352a373;rport
Max-Forwards: 70
From: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 103 BYE
User-Agent: Asterisk PBX 17.1.0
Proxy-Authorization: Digest username="usermobil", realm="asterisk", algorithm=MD5, uri="sip:ec2-18-184-51-208.eu-central-1.compute.amazonaws.com", nonce="3da3eb2a", response="a87aab29bc0dc1f244042e022c0acc2d"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
Scheduling destruction of SIP dialog 'k1-dUQgFfBN9Sl-QS4zu2A..' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:188.27.130.120:26817 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK0352a373;rport=5060;received=18.184.51.208
Contact: <sip:usermobil@188.27.130.120:26817;transport=UDP>
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=0ca6aa5b
From: <sip:100@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com>;tag=as5187003d
Call-ID: k1-dUQgFfBN9Sl-QS4zu2A..
CSeq: 103 BYE
User-Agent: Zoiper rv2.10.4.4
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'k1-dUQgFfBN9Sl-QS4zu2A..' Method: INVITE
Really destroying SIP dialog 'tKEkgBNh9V8iccn_AGGFRA..' Method: REGISTER

<--- SIP read from UDP:188.27.130.120:27087 --->

<------------->

<--- SIP read from UDP:188.27.130.120:26817 --->
REGISTER sip:ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---bbd76a0bb7d2ef38;rport
Max-Forwards: 70
Contact: <sip:usermobil@188.27.130.120:26817;transport=UDP;rinstance=80bd40cc3e0cc8e7>
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=d1ead01d
Call-ID: mPVRrIkuD67tnEopJsfGwg..
CSeq: 892 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rv2.10.4.4
Authorization: Digest username="usermobil",realm="asterisk",nonce="6cdbab29",uri="sip:ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP",response="5aa75e355c90d04f16ca2a0fa1a796bf",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 188.27.130.120:26817 (NAT)
Sending to 188.27.130.120:26817 (NAT)

<--- Transmitting (NAT) to 188.27.130.120:26817 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---bbd76a0bb7d2ef38;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=d1ead01d
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=as71674fd1
Call-ID: mPVRrIkuD67tnEopJsfGwg..
CSeq: 892 REGISTER
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="094c99b8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'mPVRrIkuD67tnEopJsfGwg..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:188.27.130.120:26817 --->
REGISTER sip:ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---b6174e898222983e;rport
Max-Forwards: 70
Contact: <sip:usermobil@188.27.130.120:26817;transport=UDP;rinstance=80bd40cc3e0cc8e7>
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=d1ead01d
Call-ID: mPVRrIkuD67tnEopJsfGwg..
CSeq: 893 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rv2.10.4.4
Authorization: Digest username="usermobil",realm="asterisk",nonce="094c99b8",uri="sip:ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP",response="38ef748ab9c557524d26fe3ca1e188c2",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 188.27.130.120:26817 (NAT)
Reliably Transmitting (NAT) to 188.27.130.120:26817:
OPTIONS sip:usermobil@188.27.130.120:26817;transport=UDP;rinstance=80bd40cc3e0cc8e7 SIP/2.0
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK69d41c7e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.31.44.215>;tag=as71599e81
To: <sip:usermobil@188.27.130.120:26817;transport=UDP;rinstance=80bd40cc3e0cc8e7>
Contact: <sip:asterisk@172.31.44.215:5060>
Call-ID: 5c5421ae199ec4ac10e0a51e2d9a0b7f@172.31.44.215:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 17.1.0
Date: Sat, 06 Jun 2020 17:55:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 188.27.130.120:26817 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:32345;branch=z9hG4bK-524287-1---b6174e898222983e;received=188.27.130.120;rport=26817
From: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=d1ead01d
To: <sip:usermobil@ec2-18-184-51-208.eu-central-1.compute.amazonaws.com;transport=UDP>;tag=as71674fd1
Call-ID: mPVRrIkuD67tnEopJsfGwg..
CSeq: 893 REGISTER
Server: Asterisk PBX 17.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:usermobil@188.27.130.120:26817;transport=UDP;rinstance=80bd40cc3e0cc8e7>;expires=60
Date: Sat, 06 Jun 2020 17:55:48 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'mPVRrIkuD67tnEopJsfGwg..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:188.27.130.120:26817 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.44.215:5060;branch=z9hG4bK69d41c7e;rport=5060;received=18.184.51.208
Contact: <sip:192.168.0.102:32345>
To: <sip:usermobil@188.27.130.120:26817;transport=UDP;rinstance=80bd40cc3e0cc8e7>;tag=b096ba1a
From: "asterisk" <sip:asterisk@172.31.44.215>;tag=as71599e81
Call-ID: 5c5421ae199ec4ac10e0a51e2d9a0b7f@172.31.44.215:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper rv2.10.4.4
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '5c5421ae199ec4ac10e0a51e2d9a0b7f@172.31.44.215:5060' Method: OPTIONS

david551 · June 6, 2020, 9:16pm

The A side hasn’t responded to the 200 OK. I assume that ```
100@172.31.44.215:5060 is the address to which it should have responded.

jcolp · June 8, 2020, 11:00am

AWS is one of the worst solutions you can use to start out with because of the network layout. This complicates things and the problem you are experiencing is quite common. To that end you have to configure Asterisk as if it was behind NAT with your public IP address, and also ensure that the firewall is open back to Asterisk.

JohnDo · June 8, 2020, 1:18pm

I configure nat=yes under each client and also tried under general but my tests are having the same results. I’m also getting the following warning in Asterisk CLI: WARNING[13182]: sip/config_parser.c:817 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead. I tried to user also nat=force_rport,comedia but still the same result.

I did the following tests:
userpc(zoiper pc)
userpc ->mizudroid -no audio, drops after 6s with the message: WARNING[13182]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 6efc57a02a96e73b24465dc625923595@172.31.44.215:5060 for seqno 103
Packet timed out after 6400ms with no response
userpc->usermobil -no audio, drops after 6s with the same messages
user pc-outuser -no audio, drops after 6s with the same messages.

usermobil (zoiper mobil)
usermobil ->outuser -one way audio from outuser to usermobil, dops oafter 6s with the message: WARNING[13182]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmissi on TTn2CxeODo0zN9Oh3xdwTw… for seqno 2
Packet timed out after 6400ms with no response
[Jun 8 12:33:12] WARNING[13182]: chan_sip.c:4143 retrans_pkt: Hanging up call TTn2CxeODo0zN9Oh3xdwTw… - no reply to our critical packet
usrmobil ->usrerpc -same as above

outuser(microSIP pc)
outuser->userpc -fine 6 seconds, dropps with WARNING[13182]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 9b41f0d27d7647888849ad3ce3e99668 for seqno 16832 (Critical Response)
– See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

mizudroid(mizudroid mobil)
mizudroid->zoiper pc -ok
mizudroid->outuser -ok, WARNING[32616][C-00000061]: res_rtp_asterisk.c:7278 ast_rtp_read: RTP Read too

I suspect there should be some settings in the accounts of the clients as for mizudroid app is working, but can’t figure out what.

jcolp · June 8, 2020, 1:21pm

You need to configure Asterisk as if it is behind NAT as well, using the “externip” and “localnet” options in sip.conf. This is because in AWS networking it is behind a NAT.

JohnDo · June 8, 2020, 2:27pm

Thank you. Under general, I added the external ip “externip” and the private IP and the mask under “externip” and this helped. I’m still troubleshooting why there is one way audio when I’m calling from zoiper on my mobile and zoiper on my laptop. Do you have any suggestions what should I check?

jcolp · June 8, 2020, 2:30pm

Did you ensure the RTP ports are also open and forwarded? Are you receiving packets? Is the location to which Asterisk is sending packets correct? (rtp set debug on).

One way audio is extremely common with NAT, so there’s tons of posts on here and other places for what to try/investigate.

JohnDo · June 8, 2020, 2:41pm

Under rtp.conf I have rtpstart=1 and rtpend=65535.
When I’m calling from zoiper to microsip, there is audio on both ways.

david551 · June 8, 2020, 3:20pm

That’s like chmod 777. You want to specify a range that is unlikely to used for anything else, and almost certainly doesn’t include the privileged sub-range.

It also didn’t answer the question, which was about the configuration of the router/firewall, not that of Asterisk.

JohnDo · June 13, 2020, 1:40pm

I have a dumb home router and I can not do any settings related to NAT or port forwarding.
But I was able to resolve the issue: on my Zoiper, I disabled “use rport for media”. Can you give me more detail why this helped to resolve the one way audio issue?

system · July 13, 2020, 1:40pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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