Asterisk WebRTC SipML5 : Remote host can't match request ACK

aymen_chetoui · December 5, 2014, 5:23pm

Hi all,
In my scenario, I am using Asterisk 13.0.1, and two SipML5 Clients that are trying to call each other.

[size=150]Configuration[/size]

Asterisk is installed in a Ubuntu VM with an IP address : 192.168.10.109
And the two client are on Chrome in Windows, the IP address is : 192.168.10.102

In sip.conf, I have :

[general]
context=public 
allowoverlap=no
realm=doubango.org
udpbindaddr=0.0.0.0:5060 
tcpenable=no 
tcpbindaddr=0.0.0.0
transport=udp,ws,wss
srvlookup=yes
nat = no 
nat = force_rport
encryption=yes
avpf=yes
force_avp=yes

dtlsenable = yes
dtlsverify = no
dtlscertfile=/etc/asterisk/keys/asterisk.pem 
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup = actpass 
dtlsfingerprint = sha-1


[1060]
allow=all
dtlsenable = yes
dtlsverify = no
dtlscertfile=/etc/asterisk/keys/asterisk.pem                
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem 
dtlssetup = actpass
dtlsfingerprint = sha-1
type=friend
username=1060
host=dynamic
secret=1060
context=default
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=yes
directmedia=no

[1061]
allow=all
dtlsenable = yes
dtlsverify = no
dtlscertfile=/etc/asterisk/keys/asterisk.pem                
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem 
dtlssetup = actpass
dtlsfingerprint = sha-1
type=friend
username=1061
host=dynamic
secret=1061
context=default
hasiax = no
hassip = yes
encryption = yes ; this
avpf = yes
icesupport = yes
videosupport=yes
directmedia=no

In the extensions.conf :

[default]
exten => 1060,1,Dial(SIP/1060)
exten => 1061,1,Dial(SIP/1061)

In the http.conf :

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088

In the rtp.conf :

[general]
rtpstart=10000
rtpend=20000
stunaddr=stun.l.google.com:19302

As for the SipML5, I used on client at Chrome default mode and the other client on the anonymous mode :
In http://sipml5.org/call.htm?svn=224# :
Display Name, Private Identity, and Password : 1060 / 1061 (as follows for Client 1 and 2)
Public Identity : sip:1060@doubango.org / sip:1061@doubango.org (as follows for Client 1 and 2)
Realm : doubango.org

In the expert mode, http://sipml5.org/expert.htm in both sides I have :
Disable Video : ticked
Enable RTCWeb Breaker : ticked
WebSocket Server URL : ws://192.168.10.109:8088/ws
ICE Servers : [{url:‘stun:stun.l.google.com:19302’}]

[size=150]Test[/size]

There is no problem when loging in the two clients, but when I try make a call from 1060 to 1061 for example :
On the client 2 side, Chrome asks to allow the micro. When I do so, the “Call is Rejected”.

Here what is happening on Asterisk Side :

<--- SIP read from WS:192.168.10.102:50162 --->
INVITE sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQMKlkFEyIu2fstl3S3DOo30PA7mNmATe;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=11ae210c967672e2981b38443f193fd4;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65358 INVITE
Content-Type: application/sdp
Content-Length: 2153
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 8287455714875358000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
m=audio 49927 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 207.162.10.166
a=rtcp:49927 IN IP4 207.162.10.166
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:3782639666 1 udp 2122194687 192.168.10.102 49927 typ host generation 0
a=candidate:3782639666 2 udp 2122194687 192.168.10.102 49927 typ host generation 0
a=candidate:771473313 1 udp 1685987071 207.162.10.166 49927 typ srflx raddr 192.168.10.102 rport 49927 generation 0
a=candidate:771473313 2 udp 1685987071 207.162.10.166 49927 typ srflx raddr 192.168.10.102 rport 49927 generation 0
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host tcptype active generation 0
a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host tcptype active generation 0
a=candidate:2952101058 1 tcp 1518214911 192.168.10.102 0 typ host tcptype active generation 0
a=candidate:2952101058 2 tcp 1518214911 192.168.10.102 0 typ host tcptype active generation 0
a=ice-ufrag:MwhBkSFG9Ov3z5qp
a=ice-pwd:rww0LRtT2qTNHirCQ/QNBlWm
a=ice-options:google-ice
a=fingerprint:sha-256 B5:27:21:87:CD:21:BF:E9:92:A9:55:16:AE:FE:5F:8E:5D:7D:4F:3C:B2:EA:CA:51:CB:D2:A1:F4:ED:03:54:E7
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
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:2380123242 cname:BsVHcarYr2rdtCpy
a=ssrc:2380123242 msid:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37 ab2994c3-e2e3-4903-b147-5eb3be8602db
a=ssrc:2380123242 mslabel:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
a=ssrc:2380123242 label:ab2994c3-e2e3-4903-b147-5eb3be8602db
<------------->
--- (12 headers 44 lines) ---
Using INVITE request as basis request - b6dd849a-c20e-c682-0802-7b0edb3365a1
Found peer '1060' for '1060' from 192.168.10.102:50162

<--- Reliably Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQMKlkFEyIu2fstl3S3DOo30PA7mNmATe;received=192.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4856ab1f
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65358 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="doubango.org", nonce="0f9f4c90"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b6dd849a-c20e-c682-0802-7b0edb3365a1' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.10.102:50162 --->
ACK sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQMKlkFEyIu2fstl3S3DOo30PA7mNmATe;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4856ab1f
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65358 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:192.168.10.102:50162 --->
INVITE sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=11ae210c967672e2981b38443f193fd4;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Content-Type: application/sdp
Content-Length: 2153
Max-Forwards: 70
Authorization: Digest username="1060",realm="doubango.org",nonce="0f9f4c90",uri="sip:1061@doubango.org",response="fad8bc7189034b543fa54058e369b451",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 8287455714875358000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
m=audio 49927 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 207.162.10.166
a=rtcp:49927 IN IP4 207.162.10.166
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 49926 typ host generation 0
a=candidate:3782639666 1 udp 2122194687 192.168.10.102 49927 typ host generation 0
a=candidate:3782639666 2 udp 2122194687 192.168.10.102 49927 typ host generation 0
a=candidate:771473313 1 udp 1685987071 207.162.10.166 49927 typ srflx raddr 192.168.10.102 rport 49927 generation 0
a=candidate:771473313 2 udp 1685987071 207.162.10.166 49927 typ srflx raddr 192.168.10.102 rport 49927 generation 0
a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host tcptype active generation 0
a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host tcptype active generation 0
a=candidate:2952101058 1 tcp 1518214911 192.168.10.102 0 typ host tcptype active generation 0
a=candidate:2952101058 2 tcp 1518214911 192.168.10.102 0 typ host tcptype active generation 0
a=ice-ufrag:MwhBkSFG9Ov3z5qp
a=ice-pwd:rww0LRtT2qTNHirCQ/QNBlWm
a=ice-options:google-ice
a=fingerprint:sha-256 B5:27:21:87:CD:21:BF:E9:92:A9:55:16:AE:FE:5F:8E:5D:7D:4F:3C:B2:EA:CA:51:CB:D2:A1:F4:ED:03:54:E7
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
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:2380123242 cname:BsVHcarYr2rdtCpy
a=ssrc:2380123242 msid:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37 ab2994c3-e2e3-4903-b147-5eb3be8602db
a=ssrc:2380123242 mslabel:7Xm9A4y63ZJIvNk5DNkisHLnRkuESOuomh37
a=ssrc:2380123242 label:ab2994c3-e2e3-4903-b147-5eb3be8602db
<------------->
--- (13 headers 44 lines) ---
Using INVITE request as basis request - b6dd849a-c20e-c682-0802-7b0edb3365a1
Found peer '1060' for '1060' from 192.168.10.102:50162
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
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 207.162.10.166:49927
Peer doesn't provide video
Looking for 1061 in default (domain doubango.org)
sip_route_dump: route/path hop: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;received=192.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1061@192.168.10.109:5060;transport=WS>
Content-Length: 0


<------------>
We think we can do text
Audio is at 16562
Video is at 192.168.10.109:19038
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding video codec h263 to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding video codec h261 to SDP
Adding video codec h263p to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding video codec vp8 to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.10.102:50163:
INVITE sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.10.109:5060;branch=z9hG4bK67ebe9be;rport
Max-Forwards: 70
From: "1060" <sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:1060@192.168.10.109:5060;transport=WS>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.0.1
Date: Fri, 05 Dec 2014 17:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1389

v=0
o=root 85583298 85583298 IN IP4 192.168.10.109
s=Asterisk PBX 13.0.1
c=IN IP4 192.168.10.109
b=CT:384
t=0 0
m=audio 16562 UDP/TLS/RTP/SAVPF 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 C5:4F:EC:21:D8:2A:23:17:02:27:83:F1:6D:D6:BE:83:AF:94:A9:B7
a=sendrecv
m=video 19038 UDP/TLS/RTP/SAVPF 34 31 98 99 104 100
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 C5:4F:EC:21:D8:2A:23:17:02:27:83:F1:6D:D6:BE:83:AF:94:A9:B7
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

---

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdnaT1vVhcoZ
Contact: <sip:1061@df7jal23ls0d.invalid;transport=ws>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1061@df7jal23ls0d.invalid;transport=ws>

<--- Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;received=192.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4715a5ca
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1061@192.168.10.109:5060;transport=WS>
Content-Length: 0


<------------>
Really destroying SIP dialog '07b1f055-deeb-64cc-0687-cb841aada377' Method: REGISTER

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdnaT1vVhcoZ
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.10.102:50163:
ACK sip:1061@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.10.109:5060;branch=z9hG4bK67ebe9be;rport
Max-Forwards: 70
From: "1060" <sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdnaT1vVhcoZ
Contact: <sip:1060@192.168.10.109:5060;transport=WS>
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.0.1
Content-Length: 0


---

<--- Reliably Transmitting (NAT) to 192.168.10.102:50162 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;received=192.168.10.102;rport=50162
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4715a5ca
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 INVITE
Server: Asterisk PBX 13.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


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

<--- SIP read from WS:192.168.10.102:50163 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 192.168.10.109:5060;rport=5060;branch=z9hG4bK67ebe9be
From: "1060"<sip:1060@192.168.10.109>;tag=as162314ad
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=Vu3uUh9OTdnaT1vVhcoZ
Call-ID: 69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
[Dec  5 12:13:51] WARNING[2865][C-00000001]: chan_sip.c:24189 handle_response: Remote host can't match request ACK to call '69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060'. Giving up.

<--- SIP read from WS:192.168.10.102:50162 --->
ACK sip:1061@doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKaRHhavVgQhNexGzeicZ856ayuCnXDk7r;rport
From: "1060"<sip:1060@doubango.org>;tag=8uEBBzWaND7t0A39IcpD
To: <sip:1061@doubango.org>;tag=as4715a5ca
Call-ID: b6dd849a-c20e-c682-0802-7b0edb3365a1
CSeq: 65359 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060' Method: INVITE
Really destroying SIP dialog 'b6dd849a-c20e-c682-0802-7b0edb3365a1' Method: INVITE
Really destroying SIP dialog '824ee07e-2f1c-e5b4-9ff8-600ba6a2d0a0' Method: REGISTER

Could you, please, help me out what went wrong and how to deal with it ?
and explain to me what does it mean :

WARNING[2865][C-00000001]: chan_sip.c:24189 handle_response: Remote host can't match request ACK to call '69af5c6775ff2046095da6a54e22d43b@192.168.10.109:5060'. Giving up.

Thanks in advance !

david55 · December 5, 2014, 5:39pm

It means Asterisk received a status 481 response.

I’d be more worried about the 603 which is where the peer starts to go seriously wrong. Whilst it should not have rejected the ACK, as it seems to have all the right identifiers, the 603 looks like an earlier internal failure.

(It is possible that a router is damaging message IDs or tags, or even sending a bogus 603.

aymen_chetoui · December 9, 2014, 2:52pm

I appreciate your response
You said that this could be an internal failure : should I reinstall Asterisk then ?

david55 · December 9, 2014, 2:59pm

An internal failure in WS:192.168.10.102:50163 not in Asterisk.

In any case, Asterisk is not Windows. Re-installing is not a universal, or even likely, panacea.