Cant' hear other side(JsSIP + Asterisk 12.1.0)

pkristel · April 18, 2014, 1:46pm

Hello all,

When I’m trying to call from web client(JsSIP) to softphone(X-Lite) via Asterisk I cannot hear anything from X-Lite to my JsIP client, in reverse its OK. I tried call between two softphone, also and its OK bothsides.

I’m using Asterisk 12.1.0 and once I’ve got this error:
[Apr 18 14:52:34] WARNING[5222][C-0000000c]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

In JsSIP mailing list think that its Asterisk issue.

Thank you for your help!

Patrik

navaismo · April 18, 2014, 1:54pm

You need to provide logs from chrome, sip debug from asterisk, a part of the rtp debug from asterisk and the sip configuration. An yes in that forum everything is wrong but their not API.

pkristel · April 18, 2014, 3:16pm

ok, when I debugg it I’ve got this Error:

I have found this kind of issue here:
issues.asterisk.org/jira/browse/ASTERISK-23026

but when I change it in /usr/local/src/asterisk-12.1.0-rc3/res/res_rtp_asterisk.c
nothing happened

I’ve still got same error…

navaismo · April 18, 2014, 3:51pm

Provide the complete requested logs.

pkristel · April 19, 2014, 1:48pm

Hi,

here’s debug -->>

sip.conf:

[general]
allowguest=yes
maxexpirey=3600
defaultexpirey=3600
port=5060
bindaddr=0.0.0.0
context=internal
language=en
dtmfmode=info


[200]                        ; JsSIP client
type=friend
secret=200
host=dynamic
context=internal
canreinvite=no
call-limit=500
nat=force_rport
transport=ws,wss
avpf=yes
encryption=yes
icesupport=yes
videosupport=no


[202]                ; X-Lite client
type=friend
secret=202
host=dynamic
context=internal
nat=yes
transport=udp
avpf=no
encryption=no
icesupport=no
videosupport=no

Chrome console debug:

JsSIP | EVENT EMITTER | emitting event registered jssip-0.3.0.min.js:9
----------registered----------- klient.html:44
JsSIP | EVENT EMITTER | adding event newDTMF jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | adding event ended jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | adding event started jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | adding event failed jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | adding event progress jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | emitting event newRTCSession jssip-0.3.0.min.js:9
JsSIP | RTC SESSION | requesting access to local media jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | got local media stream jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1560575937 1 udp 2122260223 169.254.65.8 64862 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1560575937 2 udp 2122260223 169.254.65.8 64862 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1516508954 1 udp 2122194687 192.168.0.106 64863 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1516508954 2 udp 2122194687 192.168.0.106 64863 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:3677098414 1 udp 1685987071 147.175.216.240 64863 typ srflx raddr 192.168.0.106 rport 64863 generation 0
 jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | ICE candidate received: a=candidate:3677098414 2 udp 1685987071 147.175.216.240 64863 typ srflx raddr 192.168.0.106 rport 64863 generation 0
 jssip-0.3.0.min.js:10
JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK4381869 jssip-0.3.0.min.js:9
JsSIP | DIALOG | new UAC dialog created with status EARLY jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | emitting event progress jssip-0.3.0.min.js:9
JsSIP | RTC SESSION | stream added: default jssip-0.3.0.min.js:10
JsSIP | DIALOG | dialog 6egc17kuvl2je0o87b4op8tsev60npas79d5762b  changed to CONFIRMED state jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | emitting event started jssip-0.3.0.min.js:9
JsSIP | TRANSACTION | Timer B expired for INVITE client transaction z9hG4bK1640168 jssip-0.3.0.min.js:9
JsSIP | TRANSACTION | Timer M expired for INVITE client transaction z9hG4bK1640168 jssip-0.3.0.min.js:9
JsSIP | RTC SESSION | closing INVITE session 6egc17kuvl2je0o87b4op8tsev60np jssip-0.3.0.min.js:10
JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.min.js:10
JsSIP | DIALOG | dialog 6egc17kuvl2je0o87b4op8tsev60npas79d5762b deleted jssip-0.3.0.min.js:9
JsSIP | EVENT EMITTER | emitting event ended jssip-0.3.0.min.js:9
JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK48808c0d

SIP debug:

[code]<— SIP read from WS:192.168.0.106:51709 —>
INVITE sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS hat7f7lbgktt.invalid;branch=z9hG4bK1640168
Max-Forwards: 69
To: sip:202@192.168.0.109
From: sip:200@192.168.0.109;tag=p8tsev60np
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 2789 INVITE
Authorization: Digest algorithm=MD5, username=“200”, realm=“asterisk”, nonce=“21fdd8c0”, uri="sip:202@192.168.0.109", response="077edbb83494203045c541edc89fb9e5"
Contact: sip:ugioubt2@hat7f7lbgktt.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 2150

v=0
o=- 7117681503785597723 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS kmOganHLCy2vMNaUchD012reLNNUCaUv7Dtp
m=audio 64863 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 147.175.216.240
a=rtcp:64863 IN IP4 147.175.216.240
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 64862 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 64862 typ host generation 0
a=candidate:1516508954 1 udp 2122194687 192.168.0.106 64863 typ host generation 0
a=candidate:1516508954 2 udp 2122194687 192.168.0.106 64863 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:3677098414 1 udp 1685987071 147.175.216.240 64863 typ srflx raddr 192.168.0.106 rport 64863 generation 0
a=candidate:3677098414 2 udp 1685987071 147.175.216.240 64863 typ srflx raddr 192.168.0.106 rport 64863 generation 0
a=ice-ufrag:agqGIusA/g+2fZhV
a=ice-pwd:CVzSoqyvRnAYZeqmpqyWg9Bg
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dHjkiwclb9U+ZuoKJ4+s30/i5j/nsLrWzMG3IzCF
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:J3A2MBWJgw5LgudzvuQuBm5W+zXcNk2t0ATUvlJq
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:729560918 cname:6uOjmNuEFKaaJcWh
a=ssrc:729560918 msid:kmOganHLCy2vMNaUchD012reLNNUCaUv7Dtp c7fe212f-f4b7-401c-9e9e-d47988069681
a=ssrc:729560918 mslabel:kmOganHLCy2vMNaUchD012reLNNUCaUv7Dtp
a=ssrc:729560918 label:c7fe212f-f4b7-401c-9e9e-d47988069681
<------------->
— (14 headers 45 lines) —
Using INVITE request as basis request - 6egc17kuvl2je0o87b4o
Found peer ‘200’ for ‘200’ from 192.168.0.106:51709
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 147.175.216.240:64863
Looking for 202 in internal (domain 192.168.0.109)
list_route: route/path hop: sip:ugioubt2@hat7f7lbgktt.invalid;transport=ws;ob

<— Transmitting (NAT) to 192.168.0.106:51709 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS hat7f7lbgktt.invalid;branch=z9hG4bK1640168;received=192.168.0.106;rport=51709
From: sip:200@192.168.0.109;tag=p8tsev60np
To: sip:202@192.168.0.109
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 2789 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@192.168.0.109:5060;transport=WS
Content-Length: 0

<------------>
– Executing [202@internal:1] Dial(“SIP/200-0000001e”, “SIP/202”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 16312
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Reliably Transmitting (NAT) to 192.168.0.100:29460:
INVITE sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK21cb56f9;rport
Max-Forwards: 70
From: sip:200@192.168.0.109;tag=as01d46946
To: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0
Contact: sip:200@192.168.0.109:5060
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.0-rc3
Date: Sat, 19 Apr 2014 13:39:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 1584740488 1584740488 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 16312 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

-- Called SIP/202

<— SIP read from UDP:192.168.0.100:29460 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK21cb56f9;rport=5060
To: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0
From: sip:200@192.168.0.109;tag=as01d46946
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:192.168.0.100:29460 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK21cb56f9;rport=5060
Contact: sip:202@192.168.0.100:29460
To: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0;tag=f58bda5f
From: sip:200@192.168.0.109;tag=as01d46946
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 4.5.5 stamp 71236
Allow-Events: hold, talk
Content-Length: 0

<------------->
— (10 headers 0 lines) —
list_route: route/path hop: sip:202@192.168.0.100:29460
– SIP/202-0000001f is ringing

<— Transmitting (NAT) to 192.168.0.106:51709 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS hat7f7lbgktt.invalid;branch=z9hG4bK1640168;received=192.168.0.106;rport=51709
From: sip:200@192.168.0.109;tag=p8tsev60np
To: sip:202@192.168.0.109;tag=as79d5762b
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 2789 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@192.168.0.109:5060;transport=WS
Content-Length: 0

<------------>
[Apr 19 15:39:54] ERROR[3911]: pjsip:0 <?>: icess0x2a78ce8 …Error sending STUN request: Invalid argument

<— SIP read from UDP:192.168.0.100:29460 —>

<------------->
> 0x2a59900 – Probation passed - setting RTP source address to 192.168.0.100:57248

<— SIP read from UDP:192.168.0.100:29460 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK21cb56f9;rport=5060
Contact: sip:202@192.168.0.100:29460
To: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0;tag=f58bda5f
From: sip:200@192.168.0.109;tag=as01d46946
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 160

v=0
o=- 13042388409636993 3 IN IP4 192.168.0.100
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 192.168.0.100
t=0 0
m=audio 57248 RTP/AVP 0 8
a=sendrecv
<------------->
— (12 headers 7 lines) —
Found RTP audio format 0
Found RTP audio format 8
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.100:57248
list_route: route/path hop: sip:202@192.168.0.100:29460
Transmitting (NAT) to 192.168.0.100:29460:
ACK sip:202@192.168.0.100:29460 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK56ef0865;rport
Max-Forwards: 70
From: sip:200@192.168.0.109;tag=as01d46946
To: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0;tag=f58bda5f
Contact: sip:200@192.168.0.109:5060
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.1.0-rc3
Content-Length: 0

-- SIP/202-0000001f answered SIP/200-0000001e

Audio is at 13504
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP

<— Reliably Transmitting (NAT) to 192.168.0.106:51709 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS hat7f7lbgktt.invalid;branch=z9hG4bK1640168;received=192.168.0.106;rport=51709
From: sip:200@192.168.0.109;tag=p8tsev60np
To: sip:202@192.168.0.109;tag=as79d5762b
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 2789 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@192.168.0.109:5060;transport=WS
Content-Type: application/sdp
Content-Length: 708

v=0
o=root 939550829 939550829 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 13504 RTP/SAVPF 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:12eebbb74aee5acb648bf9e202be3596
a=ice-pwd:69707c19656c86d04b258699346590b1
a=candidate:Hc0a8006d 1 UDP 2130706431 192.168.0.109 13504 typ host
a=candidate:S93afd8f0 1 UDP 1694498815 147.175.216.240 13504 typ srflx
a=candidate:Hc0a8006d 2 UDP 2130706430 192.168.0.109 13505 typ host
a=candidate:S93afd8f0 2 UDP 1694498814 147.175.216.240 13506 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:Y8vbY/3HIlHfzhpOetdYIL2rZrvzVk20qIRUf6Xw

<------------>
– Channel SIP/200-0000001e joined ‘simple_bridge’ basic-bridge <9371c6fd-8626-4d3f-ad65-9b17086a4208>
– Channel SIP/202-0000001f joined ‘simple_bridge’ basic-bridge <9371c6fd-8626-4d3f-ad65-9b17086a4208>
> 0x2a59900 – Probation passed - setting RTP source address to 192.168.0.100:57248

<— SIP read from WS:192.168.0.106:51709 —>
ACK sip:202@192.168.0.109:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS hat7f7lbgktt.invalid;branch=z9hG4bK910876
Max-Forwards: 69
To: sip:202@192.168.0.109;tag=as79d5762b
From: sip:200@192.168.0.109;tag=p8tsev60np
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 2789 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:192.168.0.100:29460 —>

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

<— SIP read from UDP:192.168.0.100:29460 —>
BYE sip:200@192.168.0.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:29460;branch=z9hG4bK-d8754z-d881f30e36d76b7d-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:202@192.168.0.100:29460
To: sip:200@192.168.0.109;tag=as01d46946
From: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0;tag=f58bda5f
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 2 BYE
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.0.100:29460 (NAT)
– Channel SIP/202-0000001f left ‘simple_bridge’ basic-bridge <9371c6fd-8626-4d3f-ad65-9b17086a4208>
Scheduling destruction of SIP dialog ‘01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060’ in 32000 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.0.100:29460 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:29460;branch=z9hG4bK-d8754z-d881f30e36d76b7d-1—d8754z-;received=192.168.0.100;rport=29460
From: sip:202@192.168.0.100:29460;rinstance=53a68afaaf01b0a0;tag=f58bda5f
To: sip:200@192.168.0.109;tag=as01d46946
Call-ID: 01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060
CSeq: 2 BYE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘01ac84b409b5e38c0563aada685d627f@192.168.0.109:5060’ Method: BYE
– Channel SIP/200-0000001e left ‘simple_bridge’ basic-bridge <9371c6fd-8626-4d3f-ad65-9b17086a4208>
== Spawn extension (internal, 202, 1) exited non-zero on 'SIP/200-0000001e’
Scheduling destruction of SIP dialog ‘6egc17kuvl2je0o87b4o’ in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.0.106:51709:
BYE sip:ugioubt2@hat7f7lbgktt.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK48808c0d;rport
Max-Forwards: 70
From: sip:202@192.168.0.109;tag=as79d5762b
To: sip:200@192.168.0.109;tag=p8tsev60np
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.1.0-rc3
Proxy-Authorization: Digest username=“cfun6j0c”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.0.109”, nonce=“21fdd8c0”, response="d7744f4d2d9caf8384148e9a4985fd31"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from WS:192.168.0.106:51709 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK48808c0d;rport
To: sip:200@192.168.0.109;tag=p8tsev60np
From: sip:202@192.168.0.109;tag=as79d5762b
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 102 BYE
Content-Length: 0

<------------->
[/code]

Thank you

navaismo · April 19, 2014, 5:11pm

The IP used in the SDP is 147.175.216.240 but the IP used in the signaling is 192.168.0.106, so if you are in the same local network check your Nat settings or use stun:null in the JSSIP API.

pkristel · April 19, 2014, 7:16pm

thanks for reply, I tried change URL of STUN server:

or:

but I’ve got same error.

and just a question: Do I need to use some turn servers? Because now JsSIP client hasn’t it in config.

var configuration ={ 'ws_servers': 'ws://192.168.0.109:8088/ws', 'uri': 'sip:200@192.168.0.109', 'password': '200', 'stun_servers': 'null' };

and if I can ask, what you exactly mean by “check NAT settings in LAN”, because all client’s, AST are in 192.168.0.0/24

navaismo · April 19, 2014, 7:47pm

[quote]and just a question: Do I need to use some turn servers? Because now JsSIP client hasn’t it in config.[/quote] Nope its not necesarry.

You have in your peer setting:

nat=force_rport

Try with:

nat=no

david55 · April 19, 2014, 8:12pm

nat=yes is deprecated in that version. You are expected to understand and use the specific options that you need.

navaismo · April 19, 2014, 11:06pm

I mean nat=no

And David the Nat settings has a lot of bugs yet also reported in JIRA so until those bugs are really fixed nat=yes works like the force_rport,comedia. Me and others have confirmed this issue since 11.3.

pkristel · April 20, 2014, 11:23pm

I’ve tried change to nat=no, in JsSIP client config: stun_servers:null, here’s SIP debug:

debian*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.0.106:41856 --->


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

<--- SIP read from WS:192.168.0.106:49583 --->
INVITE sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK2725120
Max-Forwards: 69
To: <sip:202@192.168.0.109>
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
Call-ID: 36uibb25g759ascu6sga
CSeq: 1704 INVITE
Contact: <sip:m67dvpls@240dam0nso56.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1912

v=0
o=- 2069608075674138328 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 4xzyuUd3EJTUyhapxhwlzN7vp9E3n9hRZcMI
m=audio 63897 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 169.254.65.8
a=rtcp:63897 IN IP4 169.254.65.8
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 63897 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 63897 typ host generation 0
a=candidate:1516508954 1 udp 2122194687 192.168.0.106 63898 typ host generation 0
a=candidate:1516508954 2 udp 2122194687 192.168.0.106 63898 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=ice-ufrag:jHyI+Bl+NljbzmD/
a=ice-pwd:Zgjlr07WNaCZR0LBdcgM8A/Z
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:/m6sElU9o8d/1hNBnlk1ZqGNYhPTdaJdTFeA614T
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+zYnwbPeEXbraKF2ZMak7K7XjbHDgaqfMM/uTRCa
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1279753975 cname:mgWQU05Qwz1nHwkU
a=ssrc:1279753975 msid:4xzyuUd3EJTUyhapxhwlzN7vp9E3n9hRZcMI 4574f8ed-e721-4e7a-ac14-8e9444c9c82e
a=ssrc:1279753975 mslabel:4xzyuUd3EJTUyhapxhwlzN7vp9E3n9hRZcMI
a=ssrc:1279753975 label:4574f8ed-e721-4e7a-ac14-8e9444c9c82e
<------------->
--- (13 headers 43 lines) ---
Using INVITE request as basis request - 36uibb25g759ascu6sga
Found peer '200' for '200' from 192.168.0.106:49583

<--- Reliably Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK2725120;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
To: <sip:202@192.168.0.109>;tag=as24c94902
Call-ID: 36uibb25g759ascu6sga
CSeq: 1704 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75fc950b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '36uibb25g759ascu6sga' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.0.106:49583 --->
ACK sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK2725120
To: <sip:202@192.168.0.109>;tag=as24c94902
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
Call-ID: 36uibb25g759ascu6sga
CSeq: 1704 ACK

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

<--- SIP read from WS:192.168.0.106:49583 --->
INVITE sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK4662481
Max-Forwards: 69
To: <sip:202@192.168.0.109>
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
Call-ID: 36uibb25g759ascu6sga
CSeq: 1705 INVITE
Authorization: Digest algorithm=MD5, username="200", realm="asterisk", nonce="75fc950b", uri="sip:202@192.168.0.109", response="18a0e15fb5f2f32ea7e1b7cf9bff160f"
Contact: <sip:m67dvpls@240dam0nso56.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1912

v=0
o=- 2069608075674138328 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 4xzyuUd3EJTUyhapxhwlzN7vp9E3n9hRZcMI
m=audio 63897 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 169.254.65.8
a=rtcp:63897 IN IP4 169.254.65.8
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 63897 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 63897 typ host generation 0
a=candidate:1516508954 1 udp 2122194687 192.168.0.106 63898 typ host generation 0
a=candidate:1516508954 2 udp 2122194687 192.168.0.106 63898 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=ice-ufrag:jHyI+Bl+NljbzmD/
a=ice-pwd:Zgjlr07WNaCZR0LBdcgM8A/Z
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:/m6sElU9o8d/1hNBnlk1ZqGNYhPTdaJdTFeA614T
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+zYnwbPeEXbraKF2ZMak7K7XjbHDgaqfMM/uTRCa
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1279753975 cname:mgWQU05Qwz1nHwkU
a=ssrc:1279753975 msid:4xzyuUd3EJTUyhapxhwlzN7vp9E3n9hRZcMI 4574f8ed-e721-4e7a-ac14-8e9444c9c82e
a=ssrc:1279753975 mslabel:4xzyuUd3EJTUyhapxhwlzN7vp9E3n9hRZcMI
a=ssrc:1279753975 label:4574f8ed-e721-4e7a-ac14-8e9444c9c82e
<------------->
--- (14 headers 43 lines) ---
Using INVITE request as basis request - 36uibb25g759ascu6sga
Found peer '200' for '200' from 192.168.0.106:49583
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 169.254.65.8:63897
Looking for 202 in internal (domain 192.168.0.109)
list_route: route/path hop: <sip:m67dvpls@240dam0nso56.invalid;transport=ws;ob>

<--- Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK4662481;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
To: <sip:202@192.168.0.109>
Call-ID: 36uibb25g759ascu6sga
CSeq: 1705 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202@192.168.0.109:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [202@internal:1] Dial("SIP/200-00000008", "SIP/202") in new stack
[Apr 21 00:43:21] ERROR[3572]: pjsip:0 <?>: 	icess0x2ea2af8 ..Error sending STUN request: Invalid argument
  == Using SIP RTP CoS mark 5
Audio is at 10550
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.0.100:45042:
INVITE sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK55afdaf7
Max-Forwards: 70
From: <sip:200@192.168.0.109>;tag=as07db3f45
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>
Contact: <sip:200@192.168.0.109:5060>
Call-ID: 47f7468039902be2600fe2004388683e@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.0-rc3
Date: Sun, 20 Apr 2014 22:43:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1958001771 1958001771 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 10550 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/202

<--- SIP read from UDP:192.168.0.100:45042 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK55afdaf7
Contact: <sip:202@192.168.0.100:45042>
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=33ac8923
From: <sip:200@192.168.0.109>;tag=as07db3f45
Call-ID: 47f7468039902be2600fe2004388683e@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 4.5.5 stamp 71236
Allow-Events: hold, talk
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: route/path hop: <sip:202@192.168.0.100:45042>
    -- SIP/202-00000009 is ringing

<--- Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK4662481;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
To: <sip:202@192.168.0.109>;tag=as081425fc
Call-ID: 36uibb25g759ascu6sga
CSeq: 1705 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202@192.168.0.109:5060;transport=WS>
Content-Length: 0


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

<--- SIP read from UDP:192.168.0.100:45042 --->


<------------->
       > 0x2e92670 -- Probation passed - setting RTP source address to 192.168.0.100:62900

<--- SIP read from UDP:192.168.0.100:45042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK55afdaf7
Contact: <sip:202@192.168.0.100:45042>
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=33ac8923
From: <sip:200@192.168.0.109>;tag=as07db3f45
Call-ID: 47f7468039902be2600fe2004388683e@192.168.0.109:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 160

v=0
o=- 13042507406824507 3 IN IP4 192.168.0.100
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 192.168.0.100
t=0 0
m=audio 62900 RTP/AVP 8 0
a=sendrecv
<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.100:62900
list_route: route/path hop: <sip:202@192.168.0.100:45042>
set_destination: Parsing <sip:202@192.168.0.100:45042> for address/port to send to
set_destination: set destination to 192.168.0.100:45042
Transmitting (no NAT) to 192.168.0.100:45042:
ACK sip:202@192.168.0.100:45042 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK0fafc174
Max-Forwards: 70
From: <sip:200@192.168.0.109>;tag=as07db3f45
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=33ac8923
Contact: <sip:200@192.168.0.109:5060>
Call-ID: 47f7468039902be2600fe2004388683e@192.168.0.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.1.0-rc3
Content-Length: 0


---
    -- SIP/202-00000009 answered SIP/200-00000008
Audio is at 12004
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK4662481;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
To: <sip:202@192.168.0.109>;tag=as081425fc
Call-ID: 36uibb25g759ascu6sga
CSeq: 1705 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202@192.168.0.109:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 710

v=0
o=root 2059155249 2059155249 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 12004 RTP/SAVPF 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:71500ff2252b91e60e59837d76c324ef
a=ice-pwd:42973c711c256049280a543055e3d1fd
a=candidate:Hc0a8006d 1 UDP 2130706431 192.168.0.109 12004 typ host
a=candidate:S93afd8f0 1 UDP 1694498815 147.175.216.240 12004 typ srflx
a=candidate:Hc0a8006d 2 UDP 2130706430 192.168.0.109 12005 typ host
a=candidate:S93afd8f0 2 UDP 1694498814 147.175.216.240 12006 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:GfB4ZnHSdveKNHdjQZO8V4ZNulsMV93qd+r+izFv

<------------>
    -- Channel SIP/200-00000008 joined 'simple_bridge' basic-bridge <5436fc40-353d-47ac-81b0-b726c512b9d0>
    -- Channel SIP/202-00000009 joined 'simple_bridge' basic-bridge <5436fc40-353d-47ac-81b0-b726c512b9d0>
       > 0x2e92670 -- Probation passed - setting RTP source address to 192.168.0.100:62900

<--- SIP read from WS:192.168.0.106:49583 --->
ACK sip:202@192.168.0.109:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 240dam0nso56.invalid;branch=z9hG4bK9755552
Max-Forwards: 69
To: <sip:202@192.168.0.109>;tag=as081425fc
From: <sip:200@192.168.0.109>;tag=fh7n83a83d
Call-ID: 36uibb25g759ascu6sga
CSeq: 1705 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x2ebdb20 -- Probation passed - setting RTP source address to 192.168.0.106:63898

<--- SIP read from UDP:192.168.0.100:45042 --->
BYE sip:200@192.168.0.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:45042;branch=z9hG4bK-d8754z-beba1e4b2ba2d316-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@192.168.0.100:45042>
To: <sip:200@192.168.0.109>;tag=as07db3f45
From: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=33ac8923
Call-ID: 47f7468039902be2600fe2004388683e@192.168.0.109:5060
CSeq: 2 BYE
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.0.100:45042 (no NAT)
    -- Channel SIP/202-00000009 left 'simple_bridge' basic-bridge <5436fc40-353d-47ac-81b0-b726c512b9d0>
Scheduling destruction of SIP dialog '47f7468039902be2600fe2004388683e@192.168.0.109:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.100:45042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:45042;branch=z9hG4bK-d8754z-beba1e4b2ba2d316-1---d8754z-;received=192.168.0.100;rport=45042
From: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=33ac8923
To: <sip:200@192.168.0.109>;tag=as07db3f45
Call-ID: 47f7468039902be2600fe2004388683e@192.168.0.109:5060
CSeq: 2 BYE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '47f7468039902be2600fe2004388683e@192.168.0.109:5060' Method: BYE
    -- Channel SIP/200-00000008 left 'simple_bridge' basic-bridge <5436fc40-353d-47ac-81b0-b726c512b9d0>
  == Spawn extension (internal, 202, 1) exited non-zero on 'SIP/200-00000008'
Scheduling destruction of SIP dialog '36uibb25g759ascu6sga' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:m67dvpls@240dam0nso56.invalid;transport=ws;ob> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (no NAT) to 192.168.0.106:5060:
BYE sip:m67dvpls@240dam0nso56.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK5de1263e
Max-Forwards: 70
From: <sip:202@192.168.0.109>;tag=as081425fc
To: <sip:200@192.168.0.109>;tag=fh7n83a83d
Call-ID: 36uibb25g759ascu6sga
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.1.0-rc3
Proxy-Authorization: Digest username="cm0uh2he", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.109", nonce="75fc950b", response="1249e433c86acf3c9f04e020ee1e92c5"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:192.168.0.106:49583 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK5de1263e
To: <sip:200@192.168.0.109>;tag=fh7n83a83d
From: <sip:202@192.168.0.109>;tag=as081425fc
Call-ID: 36uibb25g759ascu6sga
CSeq: 102 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '36uibb25g759ascu6sga' Method: INVITE
Really destroying SIP dialog '2d6e7bvbne70h8im6nt2lm' Method: REGISTER
debian*CLI> sip set debug off

I also try disconnect internet connection and want to see what happened without public IP, but still same error, here SIP debug also:

Really destroying SIP dialog '40ab3fb9673f1ffb187ee25552ebaf0b@192.168.0.109:5060' Method: BYE

<--- SIP read from WS:192.168.0.106:49776 --->
INVITE sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK5889935
Max-Forwards: 69
To: <sip:202@192.168.0.109>
From: <sip:200@192.168.0.109>;tag=595f9grgql
Call-ID: o4mte9kse789qceapujf
CSeq: 8753 INVITE
Contact: <sip:qbm936a0@5f8htr5upgke.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1912

v=0
o=- 4742677495389182819 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 85YBf3iA2DYGHyDkEGUwoz01sIeyl2bYsdER
m=audio 60633 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 169.254.65.8
a=rtcp:60633 IN IP4 169.254.65.8
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 60633 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 60633 typ host generation 0
a=candidate:1516508954 1 udp 2122194687 192.168.0.106 60634 typ host generation 0
a=candidate:1516508954 2 udp 2122194687 192.168.0.106 60634 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=ice-ufrag:tIXl7XUskTn49alh
a=ice-pwd:ZfD6O0GuQV8rIZvdRG6p7C7L
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:wtAnwVtkHNDW6NF+fZN9DKjW8PiUKCv0qeiYd8Ob
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:USEv4Vf5q10H3wxYFqQZ/K572unFABdezOynewD9
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2759307127 cname:LLZurREm/nx7ucAY
a=ssrc:2759307127 msid:85YBf3iA2DYGHyDkEGUwoz01sIeyl2bYsdER c88ce1c5-e16d-40c4-a1f3-948ba84a600a
a=ssrc:2759307127 mslabel:85YBf3iA2DYGHyDkEGUwoz01sIeyl2bYsdER
a=ssrc:2759307127 label:c88ce1c5-e16d-40c4-a1f3-948ba84a600a
<------------->
--- (13 headers 43 lines) ---
Using INVITE request as basis request - o4mte9kse789qceapujf
Found peer '200' for '200' from 192.168.0.106:49776

<--- Reliably Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK5889935;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=595f9grgql
To: <sip:202@192.168.0.109>;tag=as0706e3fd
Call-ID: o4mte9kse789qceapujf
CSeq: 8753 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fe9abd0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'o4mte9kse789qceapujf' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.0.106:49776 --->
ACK sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK5889935
To: <sip:202@192.168.0.109>;tag=as0706e3fd
From: <sip:200@192.168.0.109>;tag=595f9grgql
Call-ID: o4mte9kse789qceapujf
CSeq: 8753 ACK

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

<--- SIP read from WS:192.168.0.106:49776 --->
INVITE sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK1677303
Max-Forwards: 69
To: <sip:202@192.168.0.109>
From: <sip:200@192.168.0.109>;tag=595f9grgql
Call-ID: o4mte9kse789qceapujf
CSeq: 8754 INVITE
Authorization: Digest algorithm=MD5, username="200", realm="asterisk", nonce="2fe9abd0", uri="sip:202@192.168.0.109", response="cd4448db5a77e1872503409b19d39b2a"
Contact: <sip:qbm936a0@5f8htr5upgke.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1912

v=0
o=- 4742677495389182819 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 85YBf3iA2DYGHyDkEGUwoz01sIeyl2bYsdER
m=audio 60633 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 169.254.65.8
a=rtcp:60633 IN IP4 169.254.65.8
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 60633 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 60633 typ host generation 0
a=candidate:1516508954 1 udp 2122194687 192.168.0.106 60634 typ host generation 0
a=candidate:1516508954 2 udp 2122194687 192.168.0.106 60634 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=ice-ufrag:tIXl7XUskTn49alh
a=ice-pwd:ZfD6O0GuQV8rIZvdRG6p7C7L
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:wtAnwVtkHNDW6NF+fZN9DKjW8PiUKCv0qeiYd8Ob
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:USEv4Vf5q10H3wxYFqQZ/K572unFABdezOynewD9
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2759307127 cname:LLZurREm/nx7ucAY
a=ssrc:2759307127 msid:85YBf3iA2DYGHyDkEGUwoz01sIeyl2bYsdER c88ce1c5-e16d-40c4-a1f3-948ba84a600a
a=ssrc:2759307127 mslabel:85YBf3iA2DYGHyDkEGUwoz01sIeyl2bYsdER
a=ssrc:2759307127 label:c88ce1c5-e16d-40c4-a1f3-948ba84a600a
<------------->
--- (14 headers 43 lines) ---
Using INVITE request as basis request - o4mte9kse789qceapujf
Found peer '200' for '200' from 192.168.0.106:49776

<--- SIP read from UDP:192.168.0.106:41856 --->


<------------->
Really destroying SIP dialog 'iq1tt8lprg4necsn9743a6' Method: REGISTER

<--- SIP read from UDP:192.168.0.100:45042 --->


<------------->
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 169.254.65.8:60633
Looking for 202 in internal (domain 192.168.0.109)
list_route: route/path hop: <sip:qbm936a0@5f8htr5upgke.invalid;transport=ws;ob>

<--- Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK1677303;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=595f9grgql
To: <sip:202@192.168.0.109>
Call-ID: o4mte9kse789qceapujf
CSeq: 8754 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202@192.168.0.109:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [202@internal:1] Dial("SIP/200-00000010", "SIP/202") in new stack
[Apr 21 01:17:34] ERROR[3572]: pjsip:0 <?>: 	icess0x2e1c4e8 ..Error sending STUN request: Invalid argument
  == Using SIP RTP CoS mark 5
Audio is at 12410
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.0.100:45042:
INVITE sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK5698ffb0
Max-Forwards: 70
From: <sip:200@192.168.0.109>;tag=as0f73bbbd
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>
Contact: <sip:200@192.168.0.109:5060>
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.0-rc3
Date: Sun, 20 Apr 2014 23:17:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1304252665 1304252665 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 12410 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/202
Retransmitting #1 (no NAT) to 192.168.0.100:45042:
INVITE sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK5698ffb0
Max-Forwards: 70
From: <sip:200@192.168.0.109>;tag=as0f73bbbd
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>
Contact: <sip:200@192.168.0.109:5060>
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.0-rc3
Date: Sun, 20 Apr 2014 23:17:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1304252665 1304252665 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 12410 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.168.0.100:45042 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK5698ffb0
Contact: <sip:202@192.168.0.100:45042>
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=f20c3600
From: <sip:200@192.168.0.109>;tag=as0f73bbbd
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 4.5.5 stamp 71236
Allow-Events: hold, talk
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: route/path hop: <sip:202@192.168.0.100:45042>
    -- SIP/202-00000011 is ringing

<--- Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK1677303;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=595f9grgql
To: <sip:202@192.168.0.109>;tag=as10bc8f09
Call-ID: o4mte9kse789qceapujf
CSeq: 8754 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202@192.168.0.109:5060;transport=WS>
Content-Length: 0


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

<--- SIP read from UDP:192.168.0.106:41856 --->


<------------->
       > 0x2dd3110 -- Probation passed - setting RTP source address to 192.168.0.100:56366

<--- SIP read from UDP:192.168.0.100:45042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK5698ffb0
Contact: <sip:202@192.168.0.100:45042>
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=f20c3600
From: <sip:200@192.168.0.109>;tag=as0f73bbbd
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 160

v=0
o=- 13042509473689873 3 IN IP4 192.168.0.100
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 192.168.0.100
t=0 0
m=audio 56366 RTP/AVP 8 0
a=sendrecv
<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.100:56366
list_route: route/path hop: <sip:202@192.168.0.100:45042>
set_destination: Parsing <sip:202@192.168.0.100:45042> for address/port to send to
set_destination: set destination to 192.168.0.100:45042
Transmitting (no NAT) to 192.168.0.100:45042:
ACK sip:202@192.168.0.100:45042 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK3b0c8a29
Max-Forwards: 70
From: <sip:200@192.168.0.109>;tag=as0f73bbbd
To: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=f20c3600
Contact: <sip:200@192.168.0.109:5060>
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.1.0-rc3
Content-Length: 0


---
    -- SIP/202-00000011 answered SIP/200-00000010
Audio is at 13408
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.106:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK1677303;received=192.168.0.106
From: <sip:200@192.168.0.109>;tag=595f9grgql
To: <sip:202@192.168.0.109>;tag=as10bc8f09
Call-ID: o4mte9kse789qceapujf
CSeq: 8754 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202@192.168.0.109:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 566

v=0
o=root 1751935583 1751935583 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 13408 RTP/SAVPF 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:43cf99144fb02b3553ac3aa52ea26798
a=ice-pwd:4f9a570f35ef75a53f4ac45261f99d16
a=candidate:Hc0a8006d 1 UDP 2130706431 192.168.0.109 13408 typ host
a=candidate:Hc0a8006d 2 UDP 2130706430 192.168.0.109 13409 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:uqgUT/lQBpD0T/rEYxnAoVZ1+RzycethvE+qzzcF

<------------>
    -- Channel SIP/200-00000010 joined 'simple_bridge' basic-bridge <a49c7736-fb62-42ce-9295-beafdfbedcc9>
    -- Channel SIP/202-00000011 joined 'simple_bridge' basic-bridge <a49c7736-fb62-42ce-9295-beafdfbedcc9>
       > 0x2dd3110 -- Probation passed - setting RTP source address to 192.168.0.100:56366

<--- SIP read from WS:192.168.0.106:49776 --->
ACK sip:202@192.168.0.109:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 5f8htr5upgke.invalid;branch=z9hG4bK8513026
Max-Forwards: 69
To: <sip:202@192.168.0.109>;tag=as10bc8f09
From: <sip:200@192.168.0.109>;tag=595f9grgql
Call-ID: o4mte9kse789qceapujf
CSeq: 8754 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.100:45042 --->


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

<--- SIP read from UDP:192.168.0.100:45042 --->
BYE sip:200@192.168.0.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:45042;branch=z9hG4bK-d8754z-60958400e90ebe67-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@192.168.0.100:45042>
To: <sip:200@192.168.0.109>;tag=as0f73bbbd
From: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=f20c3600
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 2 BYE
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.0.100:45042 (no NAT)
Scheduling destruction of SIP dialog '5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.100:45042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:45042;branch=z9hG4bK-d8754z-60958400e90ebe67-1---d8754z-;received=192.168.0.100;rport=45042
From: <sip:202@192.168.0.100:45042;rinstance=1b0cc1af7216bb43>;tag=f20c3600
To: <sip:200@192.168.0.109>;tag=as0f73bbbd
Call-ID: 5f9b99ad3717a7c15fbf69be7c05bba4@192.168.0.109:5060
CSeq: 2 BYE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/202-00000011 left 'simple_bridge' basic-bridge <a49c7736-fb62-42ce-9295-beafdfbedcc9>
    -- Channel SIP/200-00000010 left 'simple_bridge' basic-bridge <a49c7736-fb62-42ce-9295-beafdfbedcc9>
  == Spawn extension (internal, 202, 1) exited non-zero on 'SIP/200-00000010'
Scheduling destruction of SIP dialog 'o4mte9kse789qceapujf' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:qbm936a0@5f8htr5upgke.invalid;transport=ws;ob> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (no NAT) to 192.168.0.106:5060:
BYE sip:qbm936a0@5f8htr5upgke.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK724ed4cc
Max-Forwards: 70
From: <sip:202@192.168.0.109>;tag=as10bc8f09
To: <sip:200@192.168.0.109>;tag=595f9grgql
Call-ID: o4mte9kse789qceapujf
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.1.0-rc3
Proxy-Authorization: Digest username="cm0uh2he", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.109", nonce="2fe9abd0", response="f7e31a252039512ef13a27a925a4415a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:192.168.0.106:49776 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK724ed4cc
To: <sip:200@192.168.0.109>;tag=595f9grgql
From: <sip:202@192.168.0.109>;tag=as10bc8f09
Call-ID: o4mte9kse789qceapujf
CSeq: 102 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'o4mte9kse789qceapujf' Method: INVITE
debian*CLI> sip set debug off

navaismo · April 21, 2014, 12:24am

Still wrong IP in the SDP but confirm in the RTP debug the ip used for the voice. Also check for fancy settings in your router like SIP ALG or SPI and disable it.

pkristel · April 23, 2014, 12:21pm

hey, I have tried another router(and another ISP), and the result is same, unfortunately the result is same:

[code]<------------->
— (6 headers 0 lines) —

<— SIP read from WS:192.168.0.100:50354 —>
INVITE sip:203@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS qt8gr8ekf97e.invalid;branch=z9hG4bK7342522
Max-Forwards: 69
To: sip:203@192.168.0.109
From: sip:200@192.168.0.109;tag=pcfdke95at
Call-ID: b3om425qbocj9crbtsfm
CSeq: 5925 INVITE
Authorization: Digest algorithm=MD5, username=“200”, realm=“asterisk”, nonce=“0fb76f0c”, uri="sip:203@192.168.0.109", response="4d1c8b738c8c7900aa524c8f199e5a3a"
Contact: sip:51na46gp@qt8gr8ekf97e.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 2145

v=0
o=- 261206340105167079 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS tdfzdhiY8ck6WvD5quorcqJGlNfehsX4yyKu
m=audio 63903 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 195.168.209.2
a=rtcp:63903 IN IP4 195.168.209.2
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 63902 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 63902 typ host generation 0
a=candidate:2131708102 1 udp 2122194687 192.168.0.100 63903 typ host generation 0
a=candidate:2131708102 2 udp 2122194687 192.168.0.100 63903 typ host generation 0
a=candidate:4266086002 1 udp 1685987071 195.168.209.2 63903 typ srflx raddr 192.168.0.100 rport 63903 generation 0
a=candidate:4266086002 2 udp 1685987071 195.168.209.2 63903 typ srflx raddr 192.168.0.100 rport 63903 generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:831304758 1 tcp 1518214911 192.168.0.100 0 typ host generation 0
a=candidate:831304758 2 tcp 1518214911 192.168.0.100 0 typ host generation 0
a=ice-ufrag:oDLVvj1hRwWrESg5
a=ice-pwd:VE+y6MI5KieL9iPLevsDe+Nv
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:tY5aQzHBZKv71J9X+4/tvjtGAjmrVJPtBgsoPPfR
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:c2YYGtxK/QZbkPg/eEAm1AxpopgaZCb01LJruhir
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3401872302 cname:dxlwtDcJQ3L+9d7R
a=ssrc:3401872302 msid:tdfzdhiY8ck6WvD5quorcqJGlNfehsX4yyKu 700c1600-be22-4cbb-867a-5518ca375388
a=ssrc:3401872302 mslabel:tdfzdhiY8ck6WvD5quorcqJGlNfehsX4yyKu
a=ssrc:3401872302 label:700c1600-be22-4cbb-867a-5518ca375388
<------------->
— (14 headers 45 lines) —
Using INVITE request as basis request - b3om425qbocj9crbtsfm
Found peer ‘200’ for ‘200’ from 192.168.0.100:50354
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 195.168.209.2:63903
Looking for 203 in internal (domain 192.168.0.109)
list_route: route/path hop: sip:51na46gp@qt8gr8ekf97e.invalid;transport=ws;ob

<— Transmitting (no NAT) to 192.168.0.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS qt8gr8ekf97e.invalid;branch=z9hG4bK7342522;received=192.168.0.100
From: sip:200@192.168.0.109;tag=pcfdke95at
To: sip:203@192.168.0.109
Call-ID: b3om425qbocj9crbtsfm
CSeq: 5925 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:203@192.168.0.109:5060;transport=WS
Content-Length: 0

<------------>
– Executing [203@internal:1] Dial(“SIP/200-00000022”, “SIP/203”) in new stack
[Apr 23 14:18:52] ERROR[3572]: pjsip:0 <?>: icess0x2e3f938 …Error sending STUN request: Invalid argument
== Using SIP RTP CoS mark 5
Audio is at 15214
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.0.104:50768:
INVITE sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK53052317
Max-Forwards: 70
From: sip:200@192.168.0.109;tag=as56cf7a77
To: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
Contact: sip:200@192.168.0.109:5060
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.0-rc3
Date: Wed, 23 Apr 2014 12:18:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 250

v=0
o=root 839952984 839952984 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 15214 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

-- Called SIP/203

<— SIP read from UDP:192.168.0.104:50768 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK53052317
To: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
From: sip:200@192.168.0.109;tag=as56cf7a77
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:192.168.0.104:50768 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK53052317
Contact: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
To: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP;tag=d3a21b16
From: sip:200@192.168.0.109;tag=as56cf7a77
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Zoiper r21367
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: route/path hop: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
– SIP/203-00000023 is ringing

<— Transmitting (no NAT) to 192.168.0.100:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS qt8gr8ekf97e.invalid;branch=z9hG4bK7342522;received=192.168.0.100
From: sip:200@192.168.0.109;tag=pcfdke95at
To: sip:203@192.168.0.109;tag=as2b208ade
Call-ID: b3om425qbocj9crbtsfm
CSeq: 5925 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:203@192.168.0.109:5060;transport=WS
Content-Length: 0

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

<— SIP read from UDP:192.168.0.104:50768 —>

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

<— SIP read from UDP:192.168.0.104:50768 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK53052317
Contact: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
To: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP;tag=d3a21b16
From: sip:200@192.168.0.109;tag=as56cf7a77
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r21367
Allow-Events: presence, kpml
Content-Length: 242

v=0
o=Z 0 2 IN IP4 192.168.0.104
s=Z
c=IN IP4 192.168.0.104
t=0 0
m=audio 56924 RTP/AVP 8 3 110 98 0 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 12 lines) —
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.104:56924
list_route: route/path hop: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
set_destination: Parsing sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP for address/port to send to
set_destination: set destination to 192.168.0.104:50768
Transmitting (no NAT) to 192.168.0.104:50768:
ACK sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.109:5060;branch=z9hG4bK3b7b278e
Max-Forwards: 70
From: sip:200@192.168.0.109;tag=as56cf7a77
To: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP;tag=d3a21b16
Contact: sip:200@192.168.0.109:5060
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.1.0-rc3
Content-Length: 0

-- SIP/203-00000023 answered SIP/200-00000022

Audio is at 12708
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP

<— Reliably Transmitting (no NAT) to 192.168.0.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS qt8gr8ekf97e.invalid;branch=z9hG4bK7342522;received=192.168.0.100
From: sip:200@192.168.0.109;tag=pcfdke95at
To: sip:203@192.168.0.109;tag=as2b208ade
Call-ID: b3om425qbocj9crbtsfm
CSeq: 5925 INVITE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:203@192.168.0.109:5060;transport=WS
Content-Type: application/sdp
Content-Length: 706

v=0
o=root 1996294866 1996294866 IN IP4 192.168.0.109
s=Asterisk PBX 12.1.0-rc3
c=IN IP4 192.168.0.109
t=0 0
m=audio 12708 RTP/SAVPF 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:1b04fadd3b1e788e44c93a27455e191e
a=ice-pwd:35b625a441b2247b3c59bca33b8cd728
a=candidate:Hc0a8006d 1 UDP 2130706431 192.168.0.109 12708 typ host
a=candidate:Sc3a8d102 1 UDP 1694498815 195.168.209.2 12708 typ srflx
a=candidate:Hc0a8006d 2 UDP 2130706430 192.168.0.109 12709 typ host
a=candidate:Sc3a8d102 2 UDP 1694498814 195.168.209.2 12710 typ srflx
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:SIcGmcsIa83HVI0KwHd0bEnvURyNRFNAd3pA3yXq

<------------>
– Channel SIP/200-00000022 joined ‘simple_bridge’ basic-bridge <042753aa-1f8c-456b-8ece-ceaaf37e0cc2>
– Channel SIP/203-00000023 joined ‘simple_bridge’ basic-bridge <042753aa-1f8c-456b-8ece-ceaaf37e0cc2>

<— SIP read from WS:192.168.0.100:50354 —>
ACK sip:203@192.168.0.109:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS qt8gr8ekf97e.invalid;branch=z9hG4bK8116295
Max-Forwards: 69
To: sip:203@192.168.0.109;tag=as2b208ade
From: sip:200@192.168.0.109;tag=pcfdke95at
Call-ID: b3om425qbocj9crbtsfm
CSeq: 5925 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:192.168.0.104:50768 —>
BYE sip:200@192.168.0.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-11aa0894b8e48b78-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP
To: sip:200@192.168.0.109;tag=as56cf7a77
From: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP;tag=d3a21b16
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 2 BYE
User-Agent: Zoiper r21367
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.0.104:50768 (no NAT)
Scheduling destruction of SIP dialog ‘690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.0.104:50768 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-11aa0894b8e48b78-1—d8754z-;received=192.168.0.104;rport=50768
From: sip:203@192.168.0.104:50768;rinstance=40447448fa1000a1;transport=UDP;tag=d3a21b16
To: sip:200@192.168.0.109;tag=as56cf7a77
Call-ID: 690f12d310286c1e4f902f3e6d8a45f4@192.168.0.109:5060
CSeq: 2 BYE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/203-00000023 left ‘simple_bridge’ basic-bridge <042753aa-1f8c-456b-8ece-ceaaf37e0cc2>
– Channel SIP/200-00000022 left ‘simple_bridge’ basic-bridge <042753aa-1f8c-456b-8ece-ceaaf37e0cc2>
== Spawn extension (internal, 203, 1) exited non-zero on 'SIP/200-00000022’
Scheduling destruction of SIP dialog ‘b3om425qbocj9crbtsfm’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:51na46gp@qt8gr8ekf97e.invalid;transport=ws;ob for address/port to send to
set_destination: URI is for WebSocket, we can’t set destination
Reliably Transmitting (no NAT) to 192.168.0.100:5060:
BYE sip:51na46gp@qt8gr8ekf97e.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK4a017127
Max-Forwards: 70
From: sip:203@192.168.0.109;tag=as2b208ade
To: sip:200@192.168.0.109;tag=pcfdke95at
Call-ID: b3om425qbocj9crbtsfm
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.1.0-rc3
Proxy-Authorization: Digest username=“cm0uh2he”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.0.109”, nonce=“0fb76f0c”, response="b88edacd5353fb086706a0df659f3073"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from WS:192.168.0.100:50354 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK4a017127
To: sip:200@192.168.0.109;tag=pcfdke95at
From: sip:203@192.168.0.109;tag=as2b208ade
Call-ID: b3om425qbocj9crbtsfm
CSeq: 102 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived

<— SIP read from UDP:192.168.0.104:50768 —>
PUBLISH sip:203@192.168.0.109;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-fb6dd8598f0c55a4-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.0.104:50768;transport=UDP
To: sip:203@192.168.0.109;transport=UDP
From: sip:203@192.168.0.109;transport=UDP;tag=9fac5379
Call-ID: ODQ5NGU1MDYzYzJkMjllYmU4NDdiZjA5YTg0NmI4NjE.
CSeq: 1 PUBLISH
Expires: 20
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r21367
Event: presence
Allow-Events: presence, kpml
Content-Length: 262

<?xml version="1.0" encoding="UTF-8"?>

closed Unknown

<------------->
— (16 headers 3 lines) —
Sending to 192.168.0.104:50768 (no NAT)

<— Transmitting (no NAT) to 192.168.0.104:50768 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-fb6dd8598f0c55a4-1—d8754z-;received=192.168.0.104;rport=50768
From: sip:203@192.168.0.109;transport=UDP;tag=9fac5379
To: sip:203@192.168.0.109;transport=UDP;tag=as1b6306a4
Call-ID: ODQ5NGU1MDYzYzJkMjllYmU4NDdiZjA5YTg0NmI4NjE.
CSeq: 1 PUBLISH
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘b3om425qbocj9crbtsfm’ Method: INVITE
Really destroying SIP dialog ‘ODQ5NGU1MDYzYzJkMjllYmU4NDdiZjA5YTg0NmI4NjE.’ Method: PUBLISH

<— SIP read from UDP:192.168.0.104:50768 —>
SUBSCRIBE sip:203@192.168.0.109;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-571fe38958fc6a14-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.0.104:50768;transport=UDP
To: sip:203@192.168.0.109;transport=UDP
From: sip:203@192.168.0.109;transport=UDP;tag=03d34a59
Call-ID: MjkwYjliZjY2ZWM4NGQ5MWFlZjc1MTg3YWJjMmFhYWI.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r21367
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Sending to 192.168.0.104:50768 (no NAT)
Creating new subscription
Sending to 192.168.0.104:50768 (no NAT)
list_route: route/path hop: sip:203@192.168.0.104:50768;transport=UDP
Found peer ‘203’ for ‘203’ from 192.168.0.104:50768

<— Transmitting (no NAT) to 192.168.0.104:50768 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-571fe38958fc6a14-1—d8754z-;received=192.168.0.104;rport=50768
From: sip:203@192.168.0.109;transport=UDP;tag=03d34a59
To: sip:203@192.168.0.109;transport=UDP;tag=as70de9fc3
Call-ID: MjkwYjliZjY2ZWM4NGQ5MWFlZjc1MTg3YWJjMmFhYWI.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="10a45317"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘MjkwYjliZjY2ZWM4NGQ5MWFlZjc1MTg3YWJjMmFhYWI.’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from UDP:192.168.0.104:50768 —>
SUBSCRIBE sip:203@192.168.0.109;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-795831120943b07b-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.0.104:50768;transport=UDP
To: sip:203@192.168.0.109;transport=UDP
From: sip:203@192.168.0.109;transport=UDP;tag=03d34a59
Call-ID: MjkwYjliZjY2ZWM4NGQ5MWFlZjc1MTg3YWJjMmFhYWI.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r21367
Authorization: Digest username=“203”,realm=“asterisk”,nonce=“10a45317”,uri="sip:203@192.168.0.109;transport=UDP",response=“a67265cdca42414989aecfc2f4482bf3”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 192.168.0.104:50768 (no NAT)
Found peer ‘203’ for ‘203’ from 192.168.0.104:50768

<— Transmitting (no NAT) to 192.168.0.104:50768 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.104:50768;branch=z9hG4bK-d8754z-795831120943b07b-1—d8754z-;received=192.168.0.104;rport=50768
From: sip:203@192.168.0.109;transport=UDP;tag=03d34a59
To: sip:203@192.168.0.109;transport=UDP;tag=as70de9fc3
Call-ID: MjkwYjliZjY2ZWM4NGQ5MWFlZjc1MTg3YWJjMmFhYWI.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 12.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘MjkwYjliZjY2ZWM4NGQ5MWFlZjc1MTg3YWJjMmFhYWI.’ Method: SUBSCRIBE
debian*CLI> sip set debug off
SIP Debugging Disabled
[/code]

navaismo · April 24, 2014, 3:37pm

Still differnet IP, use the null stun in your JSSIP you need to set is as:

Then try again and show us the jssip debug.

pkristel · May 3, 2014, 9:53am

I have tried ,to change it stun=nul or stun:19302=stun.l.google.com. But nothing changed. I found that the c= and a= attributes changed the IP, but that was not helpful. Pls, Can you look to JsSIP group thread on it? There is some debug.

Thank a lot!

navaismo · May 13, 2014, 10:55pm

I saw your post in the jssio forum groups.google.com/d/msg/jssip/E … 3FScTBYVMJ

And it not the “o” option is the “c” option and yes it matters, thats the ip where the RTP will go.

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