Not obtained registration for the Google account

[compnet]

Hello!

I configured Yate the instructions to connect to google voitse account. When you start Yate can not connect to the account and reports the error

 <c2s/GoogleVoice:NOTE> Dropping xml = (0x293fa1f0, stream: features) ns = http://etherx.jabber.org/streams in state = Features reason = 'required encryption not available '[0x29390700].

 Please tell me what is my mistake, OS FreeBSD.

[marian]

Hi,

Check if the openssl module is loaded.

If you built yate from svn install the openssl devel and re-build.

[compnet]

Thanks. I'm a noob, I forgot to install OpenSSL 

Now the next problem, the call is made, but no media. I know that this issue has already been discussed, does anyone know how to solve it?

<jingle/1:CALL> Outgoing. caller='IS-ALLIANCE.Supervisor@is-alliance.net/13162D82' called='14166400832@voice.google.com' [0x29a7df00]
<jingle/1:CALL> Calling. caller=IS-ALLIANCE.Supervisor@is-alliance.net/13162D82 called=14166400832@voice.google.com [0x29a7df00]
<jingle/1:NOTE> Failed to start RTP for content='jingle/1_content_1263849457' candidates local=true remote=false [0x29a7df00]
<jgengine:NOTE> Call(JG1_123598887). Sent element with id=JG1_123598887_1 confirmed by error. Terminating [0x29a92100]
<jingle/1:CALL> Redirecting to '14166400832@voice.google.com/srvenc-9a8FYoLISSeni0kafcQKDgVD9L8NMwHg' [0x29a7df00]
<jingle/2:CALL> Outgoing. caller='IS-ALLIANCE.Supervisor@is-alliance.net/13162D82' called='14166400832@voice.google.com/srvenc-9a8FYoLISSeni0kafcQKDgVD9L8NMwHg'. Transferred from=1
4166400832@voice.google.com [0x299bf400]
<jingle/2:CALL> Calling. caller=IS-ALLIANCE.Supervisor@is-alliance.net/13162D82 called=14166400832@voice.google.com/srvenc-9a8FYoLISSeni0kafcQKDgVD9L8NMwHg [0x299bf400]
<jingle/2:NOTE> Failed to start RTP for content='jingle/2_content_1074651091' candidates local=true remote=false [0x299bf400]
<jingle/1:CALL> disconnected. final=0 reason=(null) [0x29a7df00]
<jingle/1:CALL> Hangup. reason=failure [0x29a7df00]
<jingle/1:CALL> disconnected. final=1 reason=failure [0x29a7df00]
<jingle/1:CALL> Destroyed [0x29a7df00]
<NOTE> Choosing started 'audio' format 'alaw' [0x297a9c40]
<stun:NOTE> Filter: Response authenticated for '173.194.76.127:19305' - notifying RTP. [0x29a36e80]
<jingle/2:CALL> disconnected. final=0 reason=Cancelled [0x299bf400]
<jingle/2:CALL> Hangup. reason=Cancelled [0x299bf400]
<jingle/2:CALL> disconnected. final=1 reason=Cancelled [0x299bf400]
<jingle/2:CALL> Destroyed [0x299bf400]

[marian]

Hi,

In the log we can see the calling party cancelled the call.
Can't see the timing.

Can you post a log with time? (-Dt command line option)

Sent/received jabber xml output would help also (jabberclient.conf, 'general' section print-xml=yes).

[compnet]

I see in the history of calls to the Google account that the call was done, but the media is not transmitted. Maybe this is due to the NAT in VM in the cloud Amazon? Maybe need some extra settings?


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.88:5061;branch=z9hG4bK-d8e38b5c;received=80.243.7.113
From: "Yate" <sip:10070004006@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=e0172c82b60ec8e4o1
To: <sip:14166400832@ec2-54-226-217-224.compute-1.amazonaws.com>
Call-ID: 53ff535a-249d0fec@192.168.1.88
CSeq: 102 INVITE
Server: YATE/4.3.0
Content-Length: 0

------
0000010.355962 <sip/2:ALL> YateSIPConnection::YateSIPConnection(0x28fd6c90,0x2979dc00) [0x29a09400]
0000010.356058 <sip/2:ALL> NAT address is '(null)' [0x29a09400]
0000010.356268 <regfile:ALL> Authenticating user 10070004006 with password length 11
0000010.356489 <sip/2:INFO> RTP NAT detected: private '192.168.1.88' public '80.243.7.113'
0000010.356694 <INFO> Could not classify call from '10070004006', wasted 41 usec
0000010.356755 <cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/2'
0000010.356997 <INFO> Routing call to '14166400832' in context 'default' via 'jingle/14166400832@voice.google.com' in 219 usec
0000010.357034 <sip/2:ALL> NAT address is '(null)' [0x29a09400]
0000010.357349 <jingle:ALL> msgExecute. caller='IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D' called='14166400832@voice.google.com' online=true filetransfer=false
0000010.357453 <jingle/1:ALL> Jingle version set to 0 from routing
0000010.357475 <jingle/1:ALL> Session flags set to 1 from ojingle_flags=noping [0x29aa9f00]
0000010.357517 <jingle/1:CALL> Outgoing. caller='IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D' called='14166400832@voice.google.com' [0x29aa9f00]
0000010.357569 <jingle/1:CALL> Calling. caller=IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D called=14166400832@voice.google.com [0x29aa9f00]
0000010.357653 <jingle/1:ALL> Added content='jingle/1_content_1958469394' type=ice-udp initiator=true [0x29aa9f00]
0000010.357685 <jgengine:ALL> Call(JG1_80968187). Outgoing from=IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D to=14166400832@voice.google.com [0x29abe400]
0000010.358156 <jgengine:INFO> Call(JG1_80968187). Changing state from Idle to Pending [0x29abe400]
0000010.358179 <jingle/1:ALL> Using audio content 'jingle/1_content_1958469394' [0x29aa9f00]
0000010.358239 <yrtp:ALL> No-transport message received
0000010.358275 <yrtp:ALL> YRTPWrapper::YRTPWrapper('10.152.180.140',0x29aa9f00,'audio',bidir,0xbf4f02a4,false) [0x29a80900]
0000010.358295 <yrtp:ALL> YRTPWrapper::setupRTP("10.152.180.140",true) [0x29a80900]
0000010.358349 <yrtp:INFO> Session 'yrtp/1546309025' 0x298eef40 bound to 10.152.180.140:19498 +RTCP [0x29a80900]
0000010.358370 <yrtp:ALL> YRTPSource::YRTPSource(0x29a80900) [0x298eeee0]
0000010.358516 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x29a80900) [0x298eee80]
0000010.358540 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x29a80900]
0000010.359248 <jingle/1:NOTE> Failed to start RTP for content='jingle/1_content_1958469394' candidates local=true remote=false [0x29aa9f00]
0000010.416045 <jgengine:NOTE> Call(JG1_80968187). Sent element with id=JG1_80968187_1 confirmed by error. Terminating [0x29abe400]
0000010.416373 <jgengine:INFO> Call(JG1_80968187). Changing state from Pending to Ending [0x29abe400]
0000010.416393 <jgengine:INFO> Call(JG1_80968187). Changing state from Ending to Destroy [0x29abe400]
0000010.416440 <jingle/1:CALL> Redirecting to '14166400832@voice.google.com/srvenc-nW4D8j/nj1vY1zLeV32pbQ==' [0x29aa9f00]
0000010.416681 <jingle:ALL> msgExecute. caller='IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D' called='14166400832@voice.google.com/srvenc-nW4D8j/nj1vY1zLeV32pbQ==' online=true f
iletransfer=false
0000010.416750 <jingle/2:ALL> Jingle version set to 0 from routing
0000010.416769 <jingle/2:ALL> Session flags set to 1 from ojingle_flags=noping [0x29ac7400]
0000010.416812 <jingle/2:CALL> Outgoing. caller='IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D' called='14166400832@voice.google.com/srvenc-nW4D8j/nj1vY1zLeV32pbQ=='. Transferred
 from=14166400832@voice.google.com [0x29ac7400]
0000010.416857 <jingle/2:CALL> Calling. caller=IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D called=14166400832@voice.google.com/srvenc-nW4D8j/nj1vY1zLeV32pbQ== [0x29ac7400]
0000010.416889 <jingle/2:ALL> Added content='jingle/2_content_21096663' type=ice-udp initiator=true [0x29ac7400]
0000010.416912 <jgengine:ALL> Call(JG2_400318405). Outgoing from=IS-ALLIANCE.Supervisor@is-alliance.net/13A35D6D to=14166400832@voice.google.com/srvenc-nW4D8j/nj1vY1zLeV32pbQ== [0x
29a88700]
0000010.417388 <jgengine:INFO> Call(JG2_400318405). Changing state from Idle to Pending [0x29a88700]
0000010.417409 <jingle/2:ALL> Using audio content 'jingle/2_content_21096663' [0x29ac7400]
0000010.417453 <yrtp:ALL> No-transport message received
0000010.417472 <yrtp:ALL> YRTPWrapper::YRTPWrapper('10.152.180.140',0x29ac7400,'audio',bidir,0xbf591f84,false) [0x29a80700]
0000010.417489 <yrtp:ALL> YRTPWrapper::setupRTP("10.152.180.140",true) [0x29a80700]
0000010.417534 <yrtp:INFO> Session 'yrtp/943564354' 0x299fd040 bound to 10.152.180.140:19278 +RTCP [0x29a80700]
0000010.417552 <yrtp:ALL> YRTPSource::YRTPSource(0x29a80700) [0x298effa0]
0000010.417570 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x29a80700) [0x298eff40]
0000010.417585 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x29a80700]
0000010.418419 <jingle/2:NOTE> Failed to start RTP for content='jingle/2_content_21096663' candidates local=true remote=false [0x29ac7400]
0000010.418444 <jingle/1:CALL> disconnected. final=0 reason=(null) [0x29aa9f00]
0000010.418520 <jingle/1:INFO> Session terminated with reason='redirect' text='xmpp:14166400832@voice.google.com/srvenc-nW4D8j/nj1vY1zLeV32pbQ==' [0x29aa9f00]
0000010.435895 <yrtp:ALL> YRTPSource::~YRTPSource() [0x298eeee0] wrapper=0x29a80900 ts=0
0000010.435989 <yrtp:ALL> YRTPConsumer::~YRTPConsumer() [0x298eee80] wrapper=0x29a80900 ts=0
0000010.436004 <yrtp:ALL> YRTPWrapper::~YRTPWrapper() bidir 'audio' [0x29a80900]
0000010.436018 <ALL> Cleaning up RTP 0x298eef40 [0x29a80900]
0000010.436076 <jingle/1:CALL> Hangup. reason=failure [0x29aa9f00]
0000010.436091 <jingle/1:CALL> disconnected. final=1 reason=failure [0x29aa9f00]
0000010.436104 <jingle/1:CALL> Destroyed [0x29aa9f00]
0000010.455999 <jgengine:ALL> Call(JG2_400318405). Sent element with id=JG2_400318405_1 confirmed by result [0x29a88700]
0000010.475911 <yrtp:ALL> RTP/AVP message received
0000010.475967 <yrtp:INFO> Guessed local IP '10.152.180.140' for remote '80.243.7.113'
0000010.475983 <yrtp:ALL> YRTPWrapper::YRTPWrapper('10.152.180.140',0x29a09400,'audio',bidir,0x298aba60,false) [0x29a80b00]
0000010.476000 <yrtp:ALL> YRTPWrapper::setupRTP("10.152.180.140",true) [0x29a80b00]
0000010.476040 <yrtp:INFO> Session 'yrtp/1014446224' 0x298eef40 bound to 10.152.180.140:24278 +RTCP [0x29a80b00]
0000010.476058 <yrtp:ALL> YRTPSource::YRTPSource(0x29a80b00) [0x298eeee0]
0000010.476080 <INFO> DataTranslator::attachChain [0x298eeee0] '(null)' -> [0x298eff40] '(null)' not possible
0000010.476096 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x29a80b00) [0x295fffa0]
0000010.476110 <INFO> DataTranslator::attachChain [0x298effa0] '(null)' -> [0x295fffa0] '(null)' not possible
0000010.476126 <yrtp:ALL> YRTPWrapper::startRTP("80.243.7.113",16440) [0x29a80b00]
0000010.476147 <yrtp:INFO> RTP starting format 'alaw' payload 8 [0x29a80b00]
0000010.476175 >>> DataTranslator::detachChain(0x298eeee0,0x298eff40)
0000010.476190 <<< DataTranslator::detachChain
0000010.476205 <INFO> DataTranslator::attachChain [0x298eeee0] 'alaw' -> [0x298eff40] '(null)' not possible
0000010.476219 >>> DataTranslator::detachChain(0x298effa0,0x295fffa0)
0000010.476231 <<< DataTranslator::detachChain
0000010.476246 <INFO> DataTranslator::attachChain [0x298effa0] '(null)' -> [0x295fffa0] 'alaw' not possible
0000010.476391 <NOTE> Choosing started 'audio' format 'alaw' [0x29822b00]
0000010.476909 <sip:INFO> 'udp:0.0.0.0:5060' sending code 180 0x29a80a00 to 80.243.7.113:5061 [0x2848e740]
------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.88:5061;branch=z9hG4bK-d8e38b5c;received=80.243.7.113
From: "Yate" <sip:10070004006@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=e0172c82b60ec8e4o1
To: <sip:14166400832@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=2022064401
Call-ID: 53ff535a-249d0fec@192.168.1.88
CSeq: 102 INVITE
Server: YATE/4.3.0
Contact: <sip:14166400832@10.152.180.140:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 187

v=0
o=yate 1374572430 1374572430 IN IP4 10.152.180.140
s=SIP Call
c=IN IP4 10.152.180.140
t=0 0
m=audio 24278 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
------
0000010.476967 <jgengine:ALL> Call(JG2_400318405). Sent element with id=JG2_400318405_2 confirmed by result [0x29a88700]
0000010.496107 <jgengine:ALL> Call(JG2_400318405). Sent element with id=JG2_400318405_3 confirmed by result [0x29a88700]
0000010.516213 <jgengine:ALL> Call(JG2_400318405). Candidates action set to candidates [0x29a88700]
0000010.516481 <yrtp:ALL> No-transport message received
0000010.516531 <yrtp:ALL> Wrapper 0x29a80700 found by CallEndpoint 0x29ac7400
0000010.516550 <yrtp:ALL> YRTPWrapper::startRTP("173.194.76.127",19305) [0x29a80700]
0000010.516571 <yrtp:INFO> RTP starting format 'alaw' payload 8 [0x29a80700]
0000010.516594 >>> DataTranslator::detachChain(0x298effa0,0x295fffa0)
0000010.516608 <<< DataTranslator::detachChain
0000010.516627 <ALL> DataTranslator::attachChain [0x298effa0] 'alaw' -> [0x295fffa0] 'alaw' succeeded
0000010.516641 >>> DataTranslator::detachChain(0x298eeee0,0x298eff40)
0000010.516654 <<< DataTranslator::detachChain
0000010.516670 <ALL> DataTranslator::attachChain [0x298eeee0] 'alaw' -> [0x298eff40] 'alaw' succeeded
0000010.516726 <jingle/2:ALL> RTP started for content='jingle/2_content_21096663' local='10.152.180.140:19278' remote='173.194.76.127:19305' [0x29ac7400]
0000010.555932 <stun:NOTE> Filter: Response authenticated for '173.194.76.127:19305' - notifying RTP. [0x293fbf40]
0000010.575819 <yrtp:ALL> No-transport message received
0000010.575840 <yrtp:ALL> Wrapper 0x29a80700 found by ID 'yrtp/943564354'
0000010.575856 <yrtp:ALL> YRTPWrapper::startRTP("173.194.76.127",19305) [0x29a80700]
0000024.401908 <sip:INFO> 'udp:0.0.0.0:5060' received 479 bytes SIP message from 80.243.7.113:5061 [0x2848e740]
------
------
0000130.496249 <jgengine:INFO> Call(JG2_400318405). Changing state from Pending to Destroy [0x29a88700]
0000130.496292 <jingle/2:INFO> Session terminated with reason='(null)' text='(null)' [0x29ac7400]
0000130.496312 >>> DataTranslator::detachChain(0x298effa0,0x295fffa0)
0000130.496333 <<< DataTranslator::detachChain
0000130.496610 >>> DataTranslator::detachChain(0x298eeee0,0x298eff40)
0000130.496624 <<< DataTranslator::detachChain
0000130.496639 <sip/2:ALL> YateSIPConnection::disconnected() '(null)' [0x29a09400]
0000130.496690 <yrtp:ALL> YRTPSource::~YRTPSource() [0x298effa0] wrapper=0x29a80700 ts=0
0000130.496710 <yrtp:ALL> YRTPConsumer::~YRTPConsumer() [0x298eff40] wrapper=0x29a80700 ts=0
0000130.496724 <yrtp:ALL> YRTPWrapper::~YRTPWrapper() bidir 'audio' [0x29a80700]
0000130.496738 <ALL> Cleaning up RTP 0x299fd040 [0x29a80700]
0000130.496802 <jingle/2:CALL> Hangup. reason=hangup [0x29ac7400]
0000130.496817 <jingle/2:CALL> disconnected. final=1 reason=hangup [0x29ac7400]
0000130.496831 <jingle/2:CALL> Destroyed [0x29ac7400]
0000130.503504 <sip:INFO> 'udp:0.0.0.0:5060' received 478 bytes SIP message from 80.243.7.113:5061 [0x2848e740]
------
------
0000130.515937 <sip/2:ALL> YateSIPConnection::hangup() state=0 trans=0x2979dc00 error='noanswer' code=487 reason='Request Terminated' [0x29a09400]
0000130.516014 <yrtp:ALL> RTP/AVP message received
0000130.516043 <yrtp:ALL> Wrapper 0x29a80b00 found by ID 'yrtp/1014446224'
0000130.516061 <yrtp:INFO> YRTPWrapper::terminate() [0x29a80b00]
0000130.516105 <yrtp:ALL> YRTPSource::~YRTPSource() [0x298eeee0] wrapper=0x29a80b00 ts=0
0000130.516126 <yrtp:ALL> YRTPConsumer::~YRTPConsumer() [0x295fffa0] wrapper=0x29a80b00 ts=0
0000130.516139 <yrtp:ALL> YRTPWrapper::~YRTPWrapper() bidir 'audio' [0x29a80b00]
0000130.516152 <ALL> Cleaning up RTP 0x298eef40 [0x29a80b00]
0000130.516275 <sip/2:ALL> YateSIPConnection::~YateSIPConnection() [0x29a09400]
0000130.516752 <sip:INFO> 'udp:0.0.0.0:5060' sending code 487 0x29a80b00 to 80.243.7.113:5061 [0x2848e740]
------
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.88:5061;branch=z9hG4bK-d8e38b5c;received=80.243.7.113
From: "Yate" <sip:10070004006@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=e0172c82b60ec8e4o1
To: <sip:14166400832@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=2022064401
Call-ID: 53ff535a-249d0fec@192.168.1.88
CSeq: 102 INVITE
Server: YATE/4.3.0
Contact: <sip:14166400832@10.152.180.140:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
0000130.682654 <sip:INFO> 'udp:0.0.0.0:5060' received 727 bytes SIP message from 80.243.7.113:5061 [0x2848e740]
------
ACK sip:14166400832@ec2-54-226-217-224.compute-1.amazonaws.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.88:5061;branch=z9hG4bK-d8e38b5c
From: "Yate" <sip:10070004006@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=e0172c82b60ec8e4o1
To: <sip:14166400832@ec2-54-226-217-224.compute-1.amazonaws.com>;tag=2022064401
Call-ID: 53ff535a-249d0fec@192.168.1.88
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="10070004006",realm="Yate",nonce="21bbf0f3fdea3be71d501dbba2b36f48.1374572430",uri="sip:14166400832@ec2-54-226-217-224.compute-1.amazonaws.com",algor
ithm=MD5,response="872af6c9a3f2ebf62296ab6250e0873e"
Contact: "Yate" <sip:10070004006@192.168.1.88:5061>
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0

[marian]

Hi,
I noticed the calling party is behind NAT.
Maybe both yate and calling party are behind different NATs?

Check the audio on sip only:
Create the following rules in regexroute:
^12345$=tone/dial
^123456$=conf/;echo=true
Call the numbers from sip.
For the first number you should hear a dial tone. This will check audio from yate to calling party.
The second rule will create a conference with echo: it will echo the sound from calling party back. This way you can check audio in both ways.

Make sure the tonegen and conference modules are loaded.

[compnet]

I checked on the test route, media not transferred. Tell me please how to set Yate to work behind nat. Thanks.

[marian]

The issue is not yate behind NAT.
Both parties are behind different NATs.

Try the following:
In the sip listener section (I suppose is the 'general' section) set:
nat_address=your_external_ip