Different IP in SDP

Hi all,

I’m trying call to my JsSIP client from Blink. I have troubleshooted regarding WebRTC issues topic, but i’m not able to resolve issue. I have problem with audio, and I found that in SDP message from JsSIP client I have IP: 169.254.65.8, its inferface of my virtual box in W7. But when I see to my RTP flow it seems correctly. In JsSIP client I have set -> [‘stun:null’].

sip.conf

[code][general]
localnet=192.168.0.0/255.255.255.0
context=internal
allowguest=yes
transport=udp,tcp,ws
nat=no
bindaddr=0.0.0.0

[200] ; JsSIP client
secret=200
type=friend
context=internal
host=dynamic
transport=ws
directmedia=no
encryption=yes
avpf=yes
icesupport=yes
videosupport=no

[201] ; Blink client
secret=201
type=friend
context=internal
host=dynamic
transport=udp
encryption=no
avpf=no
icesupport=no
videosupport=no
[/code]

[code]<— SIP read from UDP:192.168.0.102:53705 —>
INVITE sip:200@192.168.0.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:53705;rport;branch=z9hG4bKPj9d1c34509ff444e19c1d5ba4a2ceeab1
Max-Forwards: 70
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109
Contact: sip:59148670@192.168.0.102:53705
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16423 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.8.1 (Windows)
Content-Type: application/sdp
Content-Length: 450

v=0
o=- 3608710435 3608710435 IN IP4 192.168.0.102
s=Blink 0.8.1 (Windows)
c=IN IP4 192.168.0.102
t=0 0
m=audio 50010 RTP/AVP 113 9 104 103 109 0 8 101
a=rtcp:50011
a=rtpmap:113 opus/48000
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:104 speex/32000
a=rtpmap:103 speex/16000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 19 lines) —
Sending to 192.168.0.102:53705 (no NAT)
Sending to 192.168.0.102:53705 (no NAT)
Using INVITE request as basis request - e48fceac3ce846feb5470114de7c18bd
Found peer ‘201’ for ‘201’ from 192.168.0.102:53705

<— Reliably Transmitting (no NAT) to 192.168.0.102:53705 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.102:53705;branch=z9hG4bKPj9d1c34509ff444e19c1d5ba4a2ceeab1;received=192.168.0.102;rport=53705
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109;tag=as5fa054aa
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16423 INVITE
Server: Asterisk PBX 12.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="74691c22"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘e48fceac3ce846feb5470114de7c18bd’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.0.102:53705 —>
ACK sip:200@192.168.0.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:53705;rport;branch=z9hG4bKPj9d1c34509ff444e19c1d5ba4a2ceeab1
Max-Forwards: 70
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109;tag=as5fa054aa
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16423 ACK
User-Agent: Blink 0.8.1 (Windows)
Content-Length: 0

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

<— SIP read from UDP:192.168.0.102:53705 —>
INVITE sip:200@192.168.0.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:53705;rport;branch=z9hG4bKPj0331c119eebd47df8d8df2cf3ff8af68
Max-Forwards: 70
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109
Contact: sip:59148670@192.168.0.102:53705
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16424 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.8.1 (Windows)
Authorization: Digest username=“201”, realm=“asterisk”, nonce=“74691c22”, uri="sip:200@192.168.0.109", response=“45de197219c7326aef58a9f7f03d8015”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 450

v=0
o=- 3608710435 3608710435 IN IP4 192.168.0.102
s=Blink 0.8.1 (Windows)
c=IN IP4 192.168.0.102
t=0 0
m=audio 50010 RTP/AVP 113 9 104 103 109 0 8 101
a=rtcp:50011
a=rtpmap:113 opus/48000
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:104 speex/32000
a=rtpmap:103 speex/16000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 19 lines) —
Sending to 192.168.0.102:53705 (no NAT)
Using INVITE request as basis request - e48fceac3ce846feb5470114de7c18bd
Found peer ‘201’ for ‘201’ from 192.168.0.102:53705
== Using SIP RTP CoS mark 5
Found RTP audio format 113
Found RTP audio format 9
Found RTP audio format 104
Found RTP audio format 103
Found RTP audio format 109
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format opus for ID 113
Found audio description format G722 for ID 9
Found audio description format speex for ID 104
Found audio description format speex for ID 103
Found audio description format iLBC for ID 109
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|speex16|ilbc|g722|speex32|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.102:50010
Looking for 200 in internal (domain 192.168.0.109)
list_route: route/path hop: sip:59148670@192.168.0.102:53705

<— Transmitting (no NAT) to 192.168.0.102:53705 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.102:53705;branch=z9hG4bKPj0331c119eebd47df8d8df2cf3ff8af68;received=192.168.0.102;rport=53705
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16424 INVITE
Server: Asterisk PBX 12.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@192.168.0.109:5060
Content-Length: 0

<------------>
– Executing [200@internal:1] Dial(“SIP/201-00000044”, “SIP/200”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 13420
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.103:64870:
INVITE sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK38d9a4cb
Max-Forwards: 70
From: “201” sip:201@192.168.0.109;tag=as57c9561a
To: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
Contact: sip:201@192.168.0.109:5060;transport=WS
Call-ID: 40678de17c028dc521b753c00109a8ba@192.168.0.109:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.2.0
Date: Sat, 10 May 2014 09:33:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 783

v=0
o=root 717365120 717365120 IN IP4 192.168.0.109
s=Asterisk PBX 12.2.0
c=IN IP4 192.168.0.109
t=0 0
m=audio 13420 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:534f27745bf886a95d4dfb09121d1183
a=ice-pwd:0b1ea290655a113155ba8ee0245b2a5c
a=candidate:Hc0a8006d 1 UDP 2130706431 192.168.0.109 13420 typ host
a=candidate:S93afd8f0 1 UDP 1694498815 147.175.216.240 13420 typ srflx
a=candidate:Hc0a8006d 2 UDP 2130706430 192.168.0.109 13421 typ host
a=candidate:S93afd8f0 2 UDP 1694498814 147.175.216.240 13422 typ srflx
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:K3TDbcWL4e0UhRvrwqutSZf6sihWaL2Or6nWD8gz


-- Called SIP/200

<— SIP read from WS:192.168.0.103:64870 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK38d9a4cb
To: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws;tag=r303t5efem
From: “201” sip:201@192.168.0.109;tag=as57c9561a
Call-ID: 40678de17c028dc521b753c00109a8ba@192.168.0.109:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from WS:192.168.0.103:64870 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK38d9a4cb
To: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws;tag=an99aei471
From: “201” sip:201@192.168.0.109;tag=as57c9561a
Call-ID: 40678de17c028dc521b753c00109a8ba@192.168.0.109:5060
CSeq: 102 INVITE
Contact: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
Content-Length: 0

<------------->
— (8 headers 0 lines) —
list_route: route/path hop: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
– SIP/200-00000045 is ringing

<— Transmitting (no NAT) to 192.168.0.102:53705 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.102:53705;branch=z9hG4bKPj0331c119eebd47df8d8df2cf3ff8af68;received=192.168.0.102;rport=53705
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109;tag=as58a3e04f
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16424 INVITE
Server: Asterisk PBX 12.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@192.168.0.109:5060
Content-Length: 0

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

<— SIP read from WS:192.168.0.103:64870 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK38d9a4cb
To: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws;tag=an99aei471
From: “201” sip:201@192.168.0.109;tag=as57c9561a
Call-ID: 40678de17c028dc521b753c00109a8ba@192.168.0.109:5060
CSeq: 102 INVITE
Contact: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
Content-Length: 0

<------------->
— (8 headers 0 lines) —
list_route: route/path hop: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
– SIP/200-00000045 is ringing

<— SIP read from WS:192.168.0.103:64870 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK38d9a4cb
To: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws;tag=an99aei471
From: “201” sip:201@192.168.0.109;tag=as57c9561a
Call-ID: 40678de17c028dc521b753c00109a8ba@192.168.0.109:5060
CSeq: 102 INVITE
Contact: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
Content-Type: application/sdp
Content-Length: 1369

v=0
o=- 442278383073206570 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS HKSerxwg6SPJ581F7ttGruhmDq7ULtzueT5g
m=audio 60098 RTP/SAVPF 0 8 101
c=IN IP4 169.254.65.8
a=rtcp:60100 IN IP4 169.254.65.8
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 60098 typ host generation 0
a=candidate:1840965416 1 udp 2122194687 192.168.0.103 60099 typ host generation 0
a=candidate:1560575937 2 udp 2122260222 169.254.65.8 60100 typ host generation 0
a=candidate:1840965416 2 udp 2122194686 192.168.0.103 60101 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:590945240 1 tcp 1518214911 192.168.0.103 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280446 169.254.65.8 0 typ host generation 0
a=candidate:590945240 2 tcp 1518214910 192.168.0.103 0 typ host generation 0
a=ice-ufrag:uw/rfavNthZkHPoF
a=ice-pwd:auYU+ZK8mx/PvCCUQs2vfflF
a=mid:audio
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bQH3JGX/Mkfbrejq4fbT5+ToecQNtyd8Kz/jf0FE
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:2467073893 cname:4C/WbK/kGAlQ3JGo
a=ssrc:2467073893 msid:HKSerxwg6SPJ581F7ttGruhmDq7ULtzueT5g b3f2d973-e50f-4584-897d-9aff5d04dae3
a=ssrc:2467073893 mslabel:HKSerxwg6SPJ581F7ttGruhmDq7ULtzueT5g
a=ssrc:2467073893 label:b3f2d973-e50f-4584-897d-9aff5d04dae3
<------------->
— (9 headers 28 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 169.254.65.8:60098
list_route: route/path hop: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws
[May 10 11:33:54] ERROR[12941][C-00000025]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(“3mmpsjv7jhvg.invalid”, “(null)”, …): Name or service not known
[May 10 11:33:54] WARNING[12941][C-00000025]: chan_sip.c:16170 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : '3mmpsjv7jhvg.invalid’
set_destination: Parsing sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws for address/port to send to
set_destination: URI is for WebSocket, we can’t set destination
Transmitting (no NAT) to 192.168.0.103:64870:
ACK sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.0.109:5060;branch=z9hG4bK27f57e9b
Max-Forwards: 70
From: “201” sip:201@192.168.0.109;tag=as57c9561a
To: sip:639uh0m8@3mmpsjv7jhvg.invalid;transport=ws;tag=an99aei471
Contact: sip:201@192.168.0.109:5060;transport=WS
Call-ID: 40678de17c028dc521b753c00109a8ba@192.168.0.109:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.2.0
Content-Length: 0


-- SIP/200-00000045 answered SIP/201-00000044

Audio is at 10174
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.0.102:53705 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:53705;branch=z9hG4bKPj0331c119eebd47df8d8df2cf3ff8af68;received=192.168.0.102;rport=53705
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109;tag=as58a3e04f
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16424 INVITE
Server: Asterisk PBX 12.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@192.168.0.109:5060
Content-Type: application/sdp
Content-Length: 304

v=0
o=root 1004151623 1004151623 IN IP4 192.168.0.109
s=Asterisk PBX 12.2.0
c=IN IP4 192.168.0.109
t=0 0
m=audio 10174 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
– Channel SIP/201-00000044 joined ‘simple_bridge’ basic-bridge <25c38f80-5ab4-4a7c-8717-bea12a7582f0>
– Channel SIP/200-00000045 joined ‘simple_bridge’ basic-bridge <25c38f80-5ab4-4a7c-8717-bea12a7582f0>

<— SIP read from UDP:192.168.0.102:53705 —>
ACK sip:200@192.168.0.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:53705;rport;branch=z9hG4bKPjef2589b6d02b42b1a03b0639cf084fe7
Max-Forwards: 70
From: “201” sip:201@192.168.0.109;tag=c30abcb7423a42af83f2354e952ea9e8
To: sip:200@192.168.0.109;tag=as58a3e04f
Call-ID: e48fceac3ce846feb5470114de7c18bd
CSeq: 16424 ACK
User-Agent: Blink 0.8.1 (Windows)
Content-Length: 0

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

RTP flow:

Sent RTP packet to 192.168.0.102:50012 (type 00, seq 046489, ts 1499023440, len 000160) Got RTP packet from 192.168.0.103:59923 (type 00, seq 029526, ts 1499023601, len 000160) Sent RTP packet to 192.168.0.102:50012 (type 00, seq 046490, ts 1499023600, len 000160) Got RTP packet from 192.168.0.102:50012 (type 00, seq 007917, ts 084160, len 000160)

Colud you please help me?

thank you a lot,

Patrik