Hi every one,
Incoming calls to Firefox alone working other browsers incoming call is not establishing .
In other browsers after answering the call terminating immediately.
kindly help us come out of this issue
Kindly provide protocol logging. Please remember to mark it as preformatted text, for the forum. Also provide the version of Asterisk.
Sorry david,
Actually we are trying to do WebRTC browser to browser call in that we are facing this issue.
sip level failure logs are
In this loge we are trying to connect from 6001 to 6003 with the sipml5 has the simulator
<------------>
– Executing [6003@play_annc:1] Answer(“SIP/6001-0000001f”, “”) in new stack
Audio is at 19758
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to 192.168.73.234:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKCFW70ZF8DGelLotDycb4TtQI0F39J8nA;rport;received=192.168.73.234
From: "6001"sip:6001@192.168.151.122;tag=uIDQ8RDpdc5J9ptqyBRE
To: sip:6003@192.168.151.122;tag=as740f7856
Call-ID: d1040a76-8fca-c775-ed0d-5a44f5d7d712
CSeq: 52877 INVITE
Server: Asterisk PBX 14.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:6003@192.168.151.122:5062;transport=ws
Content-Type: application/sdp
Content-Length: 821
v=0
o=root 779722921 779722921 IN IP4 192.168.151.122
s=Asterisk PBX 14.4.0
c=IN IP4 192.168.151.122
t=0 0
m=audio 19758 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:22a475dc7bd61a4521cd636c05768fc9
a=ice-pwd:40d377dd680fb25c4db9b1750074339e
a=candidate:Hc0a8977a 1 UDP 2130706431 192.168.151.122 19758 typ host
a=candidate:Hc0a86a2b 1 UDP 2130706431 192.168.106.43 19758 typ host
a=candidate:Hc0a8977a 2 UDP 2130706430 192.168.151.122 19759 typ host
a=candidate:Hc0a86a2b 2 UDP 2130706430 192.168.106.43 19759 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 C3:5A:AD:ED:60:98:32:E7:F5:74:97:13:D7:C2:79:B5:35:C8:06:16:9D:94:EC:B1:14:83:EC:FD:E8:83:03:74
a=rtcp-mux
a=sendrecv
<------------>
<— SIP read from WS:192.168.73.234:55957 —>
ACK sip:6003@192.168.151.122:5062;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKwMQO9cRdZBOOUhUpMNLX;rport
From: "6001"sip:6001@192.168.151.122;tag=uIDQ8RDpdc5J9ptqyBRE
To: sip:6003@192.168.151.122;tag=as740f7856
Contact: "6001"sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss;+g.oma.sip-im;language="en,fr"
Call-ID: d1040a76-8fca-c775-ed0d-5a44f5d7d712
CSeq: 52877 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
<------------->
— (11 headers 0 lines) —
> 0x7fe05001b3b0 – Probation passed - setting RTP source address to 192.168.73.234:51042
– Executing [6003@play_annc:2] Playback(“SIP/6001-0000001f”, “/opt/product/WebRTC/INSTALL/var/lib/asterisk/sounds/en/WelcomePrompt”) in new stack
– <SIP/6001-0000001f> Playing ‘/opt/product/WebRTC/INSTALL/var/lib/asterisk/sounds/en/WelcomePrompt.slin’ (language ‘en’)
– Executing [6003@play_annc:3] Dial(“SIP/6001-0000001f”, “SIP/6003,r,wss://192.168.151.122:9191/asterisk/ws”) in new stack
== DTLS ECDH initialized (secp256r1), faster PFS enabled
== Using SIP RTP CoS mark 5
[May 12 17:56:16] ERROR[19457][C-00000011]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“df7jal23ls0d.invalid”, “(null)”, …): Temporary failure in name resolution
[May 12 17:56:16] WARNING[19457][C-00000011]: chan_sip.c:16762 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘df7jal23ls0d.invalid’
[May 12 17:56:16] ERROR[19457][C-00000011]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
Audio is at 13958
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.73.234:56003:
INVITE sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.151.122:5062;branch=z9hG4bK12138412;rport
Max-Forwards: 70
From: “6001” sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss
Contact: sip:6001@192.168.151.122:5062;transport=ws
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 102 INVITE
User-Agent: Asterisk PBX 14.4.0
Date: Fri, 12 May 2017 12:26:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 824
v=0
o=root 1046417564 1046417564 IN IP4 192.168.151.122
s=Asterisk PBX 14.4.0
c=IN IP4 192.168.151.122
t=0 0
m=audio 13958 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:290fe9ea477d6cae4153edbc246877f0
a=ice-pwd:618b016e315b91db5fc9ff0459279b96
a=candidate:Hc0a8977a 1 UDP 2130706431 192.168.151.122 13958 typ host
a=candidate:Hc0a86a2b 1 UDP 2130706431 192.168.106.43 13958 typ host
a=candidate:Hc0a8977a 2 UDP 2130706430 192.168.151.122 13959 typ host
a=candidate:Hc0a86a2b 2 UDP 2130706430 192.168.106.43 13959 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 C3:5A:AD:ED:60:98:32:E7:F5:74:97:13:D7:C2:79:B5:35:C8:06:16:9D:94:EC:B1:14:83:EC:FD:E8:83:03:74
a=rtcp-mux
a=sendrecv
-- Called SIP/6003
<— SIP read from WS:192.168.73.234:56003 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.151.122:5062;rport=5062;branch=z9hG4bK12138412
From: "6001"sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from WS:192.168.73.234:56003 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.151.122:5062;rport=5062;branch=z9hG4bK12138412
From: "6001"sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;tag=ZpcsAsmuoGS72Ws66XTa
Contact: sips:6003@df7jal23ls0d.invalid;transport=wss
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
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: sips:6003@df7jal23ls0d.invalid;transport=wss
– SIP/6003-00000020 is ringing
<— SIP read from WS:192.168.73.234:56003 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.151.122:5062;rport=5062;branch=z9hG4bK12138412
From: "6001"sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;tag=ZpcsAsmuoGS72Ws66XTa
Contact: sips:6003@df7jal23ls0d.invalid;transport=wss
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 887
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=- 8117512065078178000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=msid-semantic: WMS AKRjRvCwZXZ8scrlScgsX6HUx7X8PReCmJ1P
m=audio 56763 UDP/TLS/RTP/SAVPF 0 101
c=IN IP4 192.168.73.234
b=AS:64
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4173113849 1 udp 2122260223 192.168.73.234 56763 typ host generation 0
a=ice-ufrag:/u/azvZghp2L8jOh
a=ice-pwd:4huWvnPH8YD10hAIelV93Sgx
a=fingerprint:sha-256 BC:BA:4F:A4:3A:C2:46:44:03:08:DF:81:9F:3B:B7:C5:89:71:7D:8B:A4:6A:05:D9:A6:EC:33:2D:D6:06:AC:46
a=setup:active
a=mid:audio
a=sendrecv
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:3426034149 cname:pKemlJ5a71BiJCT8
a=ssrc:3426034149 msid:AKRjRvCwZXZ8scrlScgsX6HUx7X8PReCmJ1P c19e4719-1abd-4e03-b712-fa4bfb0ec26b
a=ssrc:3426034149 mslabel:AKRjRvCwZXZ8scrlScgsX6HUx7X8PReCmJ1P
a=ssrc:3426034149 label:c19e4719-1abd-4e03-b712-fa4bfb0ec26b
<------------->
— (10 headers 23 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.73.234:56763
sip_route_dump: route/path hop: sips:6003@df7jal23ls0d.invalid;transport=wss
Transmitting (NAT) to 192.168.73.234:56003:
ACK sips:6003@df7jal23ls0d.invalid;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.151.122:5062;branch=z9hG4bK7b640b00;rport
Max-Forwards: 70
From: “6001” sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;tag=ZpcsAsmuoGS72Ws66XTa
Contact: sip:6001@192.168.151.122:5062;transport=ws
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 102 ACK
User-Agent: Asterisk PBX 14.4.0
Content-Length: 0
-- SIP/6003-00000020 answered SIP/6001-0000001f
Audio is at 13958
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.73.234:56003:
INVITE sips:6003@df7jal23ls0d.invalid;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.151.122:5062;branch=z9hG4bK5b4661bf;rport
Max-Forwards: 70
From: “6001” sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;tag=ZpcsAsmuoGS72Ws66XTa
Contact: sip:6001@192.168.151.122:5062;transport=ws
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 103 INVITE
User-Agent: Asterisk PBX 14.4.0
Access-URL: wss://192.168.151.122:9191/asterisk/ws;mode=active
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 824
v=0
o=root 1046417564 1046417565 IN IP4 192.168.151.122
s=Asterisk PBX 14.4.0
c=IN IP4 192.168.151.122
t=0 0
m=audio 13958 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:290fe9ea477d6cae4153edbc246877f0
a=ice-pwd:618b016e315b91db5fc9ff0459279b96
a=candidate:Hc0a8977a 1 UDP 2130706431 192.168.151.122 13958 typ host
a=candidate:Hc0a86a2b 1 UDP 2130706431 192.168.106.43 13958 typ host
a=candidate:Hc0a8977a 2 UDP 2130706430 192.168.151.122 13959 typ host
a=candidate:Hc0a86a2b 2 UDP 2130706430 192.168.106.43 13959 typ host
a=connection:new
a=setup:passive
a=fingerprint:SHA-256 C3:5A:AD:ED:60:98:32:E7:F5:74:97:13:D7:C2:79:B5:35:C8:06:16:9D:94:EC:B1:14:83:EC:FD:E8:83:03:74
a=rtcp-mux
a=sendrecv
-- Channel SIP/6003-00000020 joined 'simple_bridge' basic-bridge <ce9fb3e4-3d49-417a-afb7-f33a19cea47f>
-- Channel SIP/6001-0000001f joined 'simple_bridge' basic-bridge <ce9fb3e4-3d49-417a-afb7-f33a19cea47f>
<— SIP read from WS:192.168.73.234:56003 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.151.122:5062;rport=5062;branch=z9hG4bK5b4661bf
From: "6001"sip:6001@192.168.151.122:5062;tag=as2df08809
To: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;tag=ZpcsAsmuoGS72Ws66XTa
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 103 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from WS:192.168.73.234:56003 —>
BYE sip:6001@192.168.151.122:5062;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKB4xpSqRTS63UEKpRIxJHUvB8tNZDPZFe;rport
From: sips:6003@df7jal23ls0d.invalid;tag=ZpcsAsmuoGS72Ws66XTa
To: "6001"sip:6001@192.168.151.122:5062;tag=as2df08809
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 59253 BYE
Content-Length: 0
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
<------------->
— (14 headers 0 lines) —
Scheduling destruction of SIP dialog ‘6ee02b89261008de1a313e9976552421@192.168.151.122:5062’ in 32000 ms (Method: BYE)
<— Transmitting (NAT) to 192.168.73.234:56003 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKB4xpSqRTS63UEKpRIxJHUvB8tNZDPZFe;received=192.168.73.234;rport=56003
From: sips:6003@df7jal23ls0d.invalid;tag=ZpcsAsmuoGS72Ws66XTa
To: "6001"sip:6001@192.168.151.122:5062;tag=as2df08809
Call-ID: 6ee02b89261008de1a313e9976552421@192.168.151.122:5062
CSeq: 59253 BYE
Server: Asterisk PBX 14.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
– Channel SIP/6003-00000020 left ‘simple_bridge’ basic-bridge
– Channel SIP/6001-0000001f left ‘simple_bridge’ basic-bridge
== Spawn extension (play_annc, 6003, 3) exited non-zero on 'SIP/6001-0000001f’
Scheduling destruction of SIP dialog ‘d1040a76-8fca-c775-ed0d-5a44f5d7d712’ in 32000 ms (Method: ACK)
set_destination: Parsing sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss for address/port to send to
set_destination: URI is for WebSocket, we can’t set destination
Reliably Transmitting (no NAT) to 192.168.73.234:5060:
BYE sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.151.122:5062;branch=z9hG4bK360a11c7
Max-Forwards: 70
From: sip:6003@192.168.151.122;tag=as740f7856
To: "6001"sip:6001@192.168.151.122;tag=uIDQ8RDpdc5J9ptqyBRE
Call-ID: d1040a76-8fca-c775-ed0d-5a44f5d7d712
CSeq: 102 BYE
User-Agent: Asterisk PBX 14.4.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<— SIP read from WS:192.168.73.234:55957 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.151.122:5062;branch=z9hG4bK360a11c7
From: sip:6003@192.168.151.122;tag=as740f7856
To: "6001"sip:6001@192.168.151.122;tag=uIDQ8RDpdc5J9ptqyBRE
Contact: sips:6001@df7jal23ls0d.invalid;transport=wss
Call-ID: d1040a76-8fca-c775-ed0d-5a44f5d7d712
CSeq: 102 BYE
Content-Length: 0
<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘d1040a76-8fca-c775-ed0d-5a44f5d7d712’ Method: ACK
<— SIP read from WS:192.168.73.234:55915 —>
REGISTER sip:192.168.151.122 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKxezRaEMoqmk6yuLpncDpyO3LIcpNiasz;rport
From: "6003"sip:6003@192.168.151.122;tag=O0daoozO6S7SUJ5GGd0T
To: "6003"sip:6003@192.168.151.122
Contact: "6003"sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: cf2dfb42-a823-0bb9-bd19-14e43271e3b1
CSeq: 42569 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
<------------->
— (11 headers 0 lines) —
– Registered SIP ‘6003’ at 192.168.73.234:55915
<— Transmitting (no NAT) to 192.168.73.234:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKxezRaEMoqmk6yuLpncDpyO3LIcpNiasz;rport;received=192.168.73.234
From: "6003"sip:6003@192.168.151.122;tag=O0daoozO6S7SUJ5GGd0T
To: "6003"sip:6003@192.168.151.122;tag=as3d4b78a4
Call-ID: cf2dfb42-a823-0bb9-bd19-14e43271e3b1
CSeq: 42569 REGISTER
Server: Asterisk PBX 14.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: sips:6003@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss;expires=200
Date: Fri, 12 May 2017 12:26:22 GMT
Content-Length: 0
here we are using :
asterisk 14
chrome 58 .x
Firefox 53.x
They appear to have sent an empty Contact header, containing an invalid address (df7jal23ls0d.invalid). Seems like you didn’t use preformatted text!
The invalid domain is never going to resolve. You need to fix that in the peer. Invalid is reserved for deliberately unusable domain names.
1 Like
In the case of SIP over Websockets putting a deliberately wrong Contact is expected as the existing connection is to be reused, and once that connection goes away you can’t do anything. The chan_sip module may have a case where it’s trying to resolve it but it still sends the message.
For this you’ll need to look at the developer console in the browser that has sent the BYE to see why it did so. Asterisk is just doing as it is told.
u saved my day :). I was using jssip with asterisk 17 and just realised that the invalid domain issue was due to a missing contact header after reading your reply @david551