[Problem] Yate am Telekom-SIP: Kein Audio/RTP-Timeout

[n0emis]

Hallo,


ich würde gerne an meinem Anschluss (Telekom Magenta) die Software Yate (http://yate.ro) als Telefonanlage verwenden.

Ich habe in der accfile.conf den Anschluss wie folgt eingerichtet:

[dtag_privat]
enabled = yes
protocol = sip
description = telekom4
interval = 600
username = 0049<rufnummer>
authname = <telekom email>
password = <passwort>
registrar = tel.t-online.de
domain = tel.t-online.de
localaddress = yes

Der Verbindungsaufbau funktioniert einwandfrei, ich kann auch anrufen/angerufen werden. Allerdings kommt es hierbei nicht zu einer Übertragung von Audio, ich bekomme allerdings folgende Fehlermeldung:

20200422231152.119182 <yrtp:WARN> Initial timeout in channel sip/123 wrapper [0x7f06f8024a10]

Hier ist ein Auszug aus dem Log davor:

------
20200422231146.162474 <yrtp:ALL> RTP/AVP message received
20200422231146.162523 <yrtp:INFO> Guessed local IP '172.31.1.103' for remote '217.0.128.133'
20200422231146.162538 <yrtp:ALL> YRTPWrapper::YRTPWrapper('172.31.1.103',0x7f06f8039bb0,'audio',bidir,0x7f06f8027ba0,false) [0x7f06f8024a10]
20200422231146.162561 <yrtp:ALL> YRTPWrapper::setupRTP("172.31.1.103",true,true) [0x7f06f8024a10]
20200422231146.162595 <yrtp:INFO> Session 'yrtp/202183485' 0x7f06f803b880 bound to 172.31.1.103:22582 +RTCP [0x7f06f8024a10]
20200422231146.162608 <yrtp:ALL> YRTPSource::YRTPSource(0x7f06f8024a10) [0x7f06f80426d0]
20200422231146.162621 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x7f06f8024a10) [0x7f06f802c7b0]
20200422231146.162633 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f06f8024a10]
...
------
20200422231146.277497 <sip:INFO> 'udp:0.0.0.0:5060' sending 'INVITE sip:<angerufener>@tel.t-online.de' 0x7f070c02c060 to 217.0.128.133:5060 [0x564b19c98980]
------
INVITE sip:<angerufener>@tel.t-online.de SIP/2.0
Max-Forwards: 19
Via: SIP/2.0/UDP 84.140.106.3:44420;rport;branch=z9hG4bK1583462757
From: "telekom4" <sip:<rufnummer>@tel.t-online.de>;tag=1821790199
To: <sip:<angerufener>@tel.t-online.de>
Call-ID: [email protected]
User-Agent: YATE/6.2.1
Contact: <sip:<rufnummer>@84.140.106.3:44420>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
CSeq: 73 INVITE
Proxy-Authorization: Digest username="<telekom email>", realm="tel.t-online.de", nonce="<nonce>", uri="sip:<angerufener>@tel.t-online.de", response="<response>", algorithm=MD5, qop=auth, nc=00000052, cnonce="<cnonce>"
Content-Type: application/sdp
Content-Length: 312

v=0
o=yate 1587597106 1587597106 IN IP4 84.140.106.3
s=SIP Call
c=IN IP4 84.140.106.3
t=0 0
m=audio 22582 RTP/AVP 8 0 4 98 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
------
20200422231146.494221 <sip:INFO> 'udp:0.0.0.0:5060' received 299 bytes SIP message from 217.0.128.133:5060 [0x564b19c98980]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.140.106.3:44420;received=84.140.106.3;rport=44420;branch=z9hG4bK1583462757
To: <sip:<angerufener>@tel.t-online.de>
From: "telekom4" <sip:<rufnummer>@tel.t-online.de>;tag=1821790199
Call-ID: [email protected]
CSeq: 73 INVITE
Content-Length: 0

------
20200422231149.114977 <sip:INFO> 'udp:0.0.0.0:5060' received 838 bytes SIP message from 217.0.128.133:5060 [0x564b19c98980]
------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 84.140.106.3:44420;received=84.140.106.3;rport=44420;branch=z9hG4bK1583462757
To: <sip:<angerufener>@tel.t-online.de>;tag=h7g4Esbg_p65542t1587597106m336760c158791039s1_1100077273-779448790
From: "telekom4" <sip:<rufnummer>@tel.t-online.de>;tag=1821790199
Call-ID: [email protected]
CSeq: 73 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.128.133;transport=udp;lr>
P-Early-Media: sendrecv
Supported: timer
Content-Type: application/sdp
Content-Length: 207
Allow: UPDATE, REFER, PRACK, NOTIFY, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER

v=0
o=- 372824712 1102818990 IN IP4 217.0.128.133
s=-
c=IN IP4 217.0.7.69
t=0 0
m=audio 34858 RTP/AVP 9 101
b=AS:80
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:30
------
20200422231149.115509 <INFO> Choosing offered 'audio' format 'g722/16000' [0x7f06f8028710]
20200422231149.115558 <sip/123:ALL> Set media: audio=g722/16000 [0x7f06f8039bb0]
20200422231149.115709 <yrtp:ALL> RTP/AVP message received
20200422231149.115725 <yrtp:ALL> Wrapper 0x7f06f8024a10 found by CallEndpoint 0x7f06f8039bb0
20200422231149.115738 <yrtp:ALL> YRTPWrapper::startRTP("217.0.7.69",34858) [0x7f06f8024a10]
20200422231149.115751 <yrtp:INFO> RTP starting format 'g722/16000' payload 9 [0x7f06f8024a10]
20200422231149.117311 <INFO> Choosing offered 'audio' format 'g722/16000' [0x7f06f8030c40]
20200422231149.117346 <sip/122:NOTE> Formats for 'audio' changed to 'g722/16000' [0x7f06f801c460]
20200422231149.117398 <yrtp:ALL> RTP/AVP message received
20200422231149.117445 <yrtp:INFO> Guessed local IP '192.168.1.206' for remote '192.168.6.2'
20200422231149.117458 <yrtp:ALL> YRTPWrapper::YRTPWrapper('192.168.1.206',0x7f06f801c460,'audio',bidir,0x564b19bd0060,false) [0x7f0708022f00]
20200422231149.117471 <yrtp:ALL> YRTPWrapper::setupRTP("192.168.1.206",true,true) [0x7f0708022f00]
20200422231149.117502 <yrtp:INFO> Session 'yrtp/1517881392' 0x7f06f801ce40 bound to 192.168.1.206:27894 +RTCP [0x7f0708022f00]
20200422231149.117516 <yrtp:ALL> YRTPSource::YRTPSource(0x7f0708022f00) [0x7f070400cb10]
20200422231149.117528 <INFO> DataTranslator::attachChain [0x7f070400cb10] '(null)' -> [0x7f06f802c7b0] 'g722/16000' not possible
20200422231149.117540 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x7f0708022f00) [0x7f0708004f30]
20200422231149.117550 <INFO> DataTranslator::attachChain [0x7f06f80426d0] 'g722/16000' -> [0x7f0708004f30] '(null)' not possible
20200422231149.117561 <yrtp:ALL> YRTPWrapper::startRTP("192.168.6.2",5004) [0x7f0708022f00]
20200422231149.117573 <yrtp:INFO> RTP starting format 'g722/16000' payload 9 [0x7f0708022f00]
20200422231149.117587 >>> DataTranslator::detachChain(0x7f070400cb10,0x7f06f802c7b0)
20200422231149.117597 <<< DataTranslator::detachChain
20200422231149.117609 <ALL> DataTranslator::attachChain [0x7f070400cb10] 'g722/16000' -> [0x7f06f802c7b0] 'g722/16000' succeeded
20200422231149.117620 >>> DataTranslator::detachChain(0x7f06f80426d0,0x7f0708004f30)
20200422231149.117629 <<< DataTranslator::detachChain
20200422231149.117641 <ALL> DataTranslator::attachChain [0x7f06f80426d0] 'g722/16000' -> [0x7f0708004f30] 'g722/16000' succeeded

Interessanterweise tun mein SipGate und mein Eventphone-Anschluss einwandfrei...

Haben hier eventuell schonmal Menschen erfolgreich Yate an einem Telekom-SIP-Anschluss zum laufen bekommen oder haben eine Idee, was das Problem sein könnte?


Vielen Dank im Vorraus...

 

[Meester Proper]

Stimmt denn die IP-Adresse aus der c-line im INVITE mit deiner "echten" IP-Adresse überein? Wie sieht ein PCAP aus, gibt es kein eingehendes RTP, nur ausgehendes?