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SIP trunk with SRTP

laurent

By laurent
in Trunk Setup

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laurent Newbie

laurent

Posted
Posted

Dear all,

 

I try to setup a trunk from a snomone to an audiocode M1000 with TLS and SRTP.

 

both element are not on the same network, the M1000 has a public IP on internet and the snomone is behind a NAT with a private IP

 

 

I have defined a SIP proxy with the IP of the M1000 as proxy (I'm doing IP authentication so no Register)

 

the TLs part is OK but I have a problem with RTP/SRTP. I have voice only in one way (only outgoing voice is ok).

 

I have think about a NAT issue but it's not the case as I see RTP in both way between the audiocode and snomone PBX (trace done on snomone server) but I see that snomone is not forwarding the RTP to the final phone.

 

I have checked the log and I see that snomone has blocked RTP : Dropped 1000 SRTP packets with wrong MAC

 

how can I solve this issue ?

 

Best regards

 

 

 

 

Laurent

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laurent Newbie

laurent

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Posted

and for Incoming calls,

as you can see in the log bellow the M1000 propose the SRTP option in the INVITE SDP but in the 200 OK sendback from the Snomone PBX to the M1000 they is no SRTP proposition.

(for incoming calls RTP is working in both way)

 

any idea how to force snomone to use SRTP ?

 

Laurent

 



[8] 2012/07/09 12:23:46:

Last message repeated 2 times



[5] 2012/07/09 12:23:46:

SIP Rx tls:95.128.80.120:54209:



INVITE sip:96956110529@95.128.80.120 SIP/2.0
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias
Max-Forwards: 10
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120
To: <sip:0325520053@95.128.80.120>
Call-ID: 1200926091972012122345@95.128.80.120
CSeq: 1 INVITE
Contact: <sip:96956110529@95.128.80.120:5067;transport=tls>
Allow: ACK,CANCEL,BYE,INFO
User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001
Content-Type: application/sdp
Content-Length: 759
x-changeuri: 1

v=0
o=Dialogic_SDP 1200505965 1200505930 IN IP4 95.128.80.120
s=Dialogic-SIP
c=IN IP4 95.128.80.120
t=0 0
m=audio 8070 RTP/AVP 8 0 98 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:98 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
m=audio 8070 RTP/SAVP 8 0 98 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:98 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pmP4qAsGAw1ojK/EYz6yU076pauMb79BPc95T+iw|2^31|3:1
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:1MkcGr7NlCSAXuecmu3HCWuZUtbfoTfqDqAGkOPX|2^31|82:1




[8] 2012/07/09 12:23:46:

Allocating for call port 23, SIP call id 1200926091972012122345@95.128.80.120 



[9] 2012/07/09 12:23:46:

UDP(IPv4): Opening socket on 0.0.0.0:58306



[9] 2012/07/09 12:23:46:

UDP(IPv4): Opening socket on 0.0.0.0:58307



[9] 2012/07/09 12:23:46:

UDP(IPv6): Opening socket on [::]:58306



[9] 2012/07/09 12:23:46:

UDP(IPv6): Opening socket on [::]:58307



[5] 2012/07/09 12:23:46:

Identify trunk (IP address and domain match) 2



[5] 2012/07/09 12:23:46:

SIP Tx tls:95.128.80.120:54209:



SIP/2.0 100 Trying
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120
To: <sip:0325520053@95.128.80.120>;tag=096ec469b7
Call-ID: 1200926091972012122345@95.128.80.120
CSeq: 1 INVITE
Content-Length: 0





[6] 2012/07/09 12:23:46:

Call-leg 23: Sending RTP for 1200926091972012122345@95.128.80.120 to 95.128.80.120:8070, codec not set yet



[8] 2012/07/09 12:23:46:

Incoming call: Request URI sip:96956110529@95.128.80.120, To is <sip:0325520053@95.128.80.120>



[8] 2012/07/09 12:23:46:

Call from a trunk 2



[8] 2012/07/09 12:23:46:

Trunk Peoplefone@pbx.company.com has country code not set, area code not set



[9] 2012/07/09 12:23:46:

Incoming: formatted From is = "0763770377" <sip:+0763770377@95.128.80.91;user=phone>



[9] 2012/07/09 12:23:46:

Incoming: formatted To is = <sip:0325520053@95.128.80.120;user=phone>



[9] 2012/07/09 12:23:46:

Incoming: formatted URI is = sip:96956110529@pbx.company.com;user=phone



[8] 2012/07/09 12:23:46:

To is <sip:0325520053@95.128.80.120;user=phone>, user 0, domain 1



[8] 2012/07/09 12:23:46:

Send call to extension ERE returned 40



[5] 2012/07/09 12:23:46:

Domain trunk Peoplefone@pbx.company.com sends call to 40 in domain pbx.company.com



[8] 2012/07/09 12:23:46:

Set the To domain based on To user 40@pbx.company.com



[8] 2012/07/09 12:23:46:

Call state for call object 13: idle



[7] 2012/07/09 12:23:46:

Call port 23: set_codecs for 1200926091972012122345@95.128.80.120 codecs "", codec_preference count 6



[8] 2012/07/09 12:23:46:

Call state for call object 13: alerting



[8] 2012/07/09 12:23:46:

Play audio_moh/noise.wav, caching true



[8] 2012/07/09 12:23:46:

Allocating for call port 24, SIP call id c580351d@pbx 



[9] 2012/07/09 12:23:46:

UDP(IPv4): Opening socket on 0.0.0.0:50840



[9] 2012/07/09 12:23:46:

UDP(IPv4): Opening socket on 0.0.0.0:50841



[9] 2012/07/09 12:23:46:

UDP(IPv6): Opening socket on [::]:50840



[9] 2012/07/09 12:23:46:

UDP(IPv6): Opening socket on [::]:50841



[7] 2012/07/09 12:23:46:

Call port 24: set_codecs for c580351d@pbx codecs "", codec_preference count 6



[9] 2012/07/09 12:23:46:

Using outbound proxy sip:192.168.1.41:4041;transport=tls because of flow-label



[8] 2012/07/09 12:23:46:

call port 24: state code from 0 to 100



[9] 2012/07/09 12:23:46:

Call port 24: update_codecs for c580351d@pbx: adding codec pcmu/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 24: update_codecs for c580351d@pbx: adding codec pcma/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 24: update_codecs for c580351d@pbx: adding codec g722/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 24: update_codecs for c580351d@pbx: adding codec g726-32/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 24: update_codecs for c580351d@pbx: adding codec gsm/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 24: update_codecs for c580351d@pbx: codec_preference size 6, available codecs size 6



[5] 2012/07/09 12:23:46:

SIP Tx tls:192.168.1.41:4041:



INVITE sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>
Call-ID: c580351d@pbx
CSeq: 1774 INVITE
Max-Forwards: 70
Contact: <sip:40@192.168.1.201:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Alert-Info: <http://127.0.0.1/Bellcore-dr3>
Content-Type: application/sdp
Content-Length: 423

v=0
o=- 1293832005 1293832005 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 50840 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hyK1Q5IAZUcPpSQujETIKGmweDxVup6NqNjBQHsR
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[8] 2012/07/09 12:23:46:

call port 23: state code from 0 to 100



[9] 2012/07/09 12:23:46:

Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: adding codec pcmu/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: adding codec pcma/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: Other side does not support codec g722/8000



[9] 2012/07/09 12:23:46:

Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: adding codec g726-32/8000 to available list



[9] 2012/07/09 12:23:46:

Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: Other side does not support codec gsm/8000



[9] 2012/07/09 12:23:46:

Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: codec_preference size 6, available codecs size 4



[5] 2012/07/09 12:23:46:

SIP Rx tls:192.168.1.41:4041:



SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr
Call-ID: c580351d@pbx
CSeq: 1774 INVITE
Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1
Content-Length: 0





[9] 2012/07/09 12:23:46:

Message repetition, packet dropped



[5] 2012/07/09 12:23:46:

SIP Rx tls:192.168.1.41:4041:



SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr
Call-ID: c580351d@pbx
CSeq: 1774 INVITE
Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1
Require: 100rel
RSeq: 1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0





[5] 2012/07/09 12:23:46:

SIP Tx tls:192.168.1.41:4041:



PRACK sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-9005154d23fdbcef188a71ccd3fc2287;rport
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr
Call-ID: c580351d@pbx
CSeq: 1775 PRACK
Max-Forwards: 70
Contact: <sip:40@192.168.1.201:5061;transport=tls>
RAck: 1 1774 INVITE
Content-Length: 0





[8] 2012/07/09 12:23:46:

Play audio_en/ringback.wav, caching true



[8] 2012/07/09 12:23:46:

call port 23: state code from 100 to 183



[6] 2012/07/09 12:23:46:

Call-leg 23: Codec pcmu/8000 is chosen for call id 1200926091972012122345@95.128.80.120



[5] 2012/07/09 12:23:46:

set codec: codec pcmu/8000 is set to call-leg 23



[5] 2012/07/09 12:23:46:

SIP Tx tls:95.128.80.120:54209:



SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120
To: <sip:0325520053@95.128.80.120>;tag=096ec469b7
Call-ID: 1200926091972012122345@95.128.80.120
CSeq: 1 INVITE
Contact: <sip:99999@192.168.1.201:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1705764368 1705764368 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 58306 RTP/AVP 0 8 98 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[5] 2012/07/09 12:23:46:

SIP Rx tls:192.168.1.41:4041:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-9005154d23fdbcef188a71ccd3fc2287;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr
Call-ID: c580351d@pbx
CSeq: 1775 PRACK
Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1
Content-Length: 0





[7] 2012/07/09 12:23:46:

Call c580351d@pbx: Clear last request



[5] 2012/07/09 12:23:48:

SIP Rx tls:192.168.1.41:4041:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr
Call-ID: c580351d@pbx
CSeq: 1774 INVITE
Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1
User-Agent: snom821/8.4.35
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 439

v=0
o=root 761033547 761033548 IN IP4 192.168.1.41
s=call
c=IN IP4 192.168.1.41
t=0 0
m=audio 53224 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:gthWqnKybO3EFXOfhrP0Y2ubFNPmXed6fqdCCVEt
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv




[7] 2012/07/09 12:23:48:

Call c580351d@pbx: Clear last INVITE



[6] 2012/07/09 12:23:48:

Call-leg 24: Codec pcmu/8000 is chosen for call id c580351d@pbx



[6] 2012/07/09 12:23:48:

Call-leg 24: Sending RTP for c580351d@pbx to 192.168.1.41:53224, codec pcmu/8000



[5] 2012/07/09 12:23:48:

set codec: codec pcmu/8000 is set to call-leg 24



[5] 2012/07/09 12:23:48:

SIP Tx tls:192.168.1.41:4041:



ACK sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-acfd9fd54aa58f898d9bca5a8194a2c9;rport
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984
To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr
Call-ID: c580351d@pbx
CSeq: 1774 ACK
Max-Forwards: 70
Contact: <sip:40@192.168.1.201:5061;transport=tls>
Content-Length: 0





[7] 2012/07/09 12:23:48:

Determine pass-through mode after receiving response



[8] 2012/07/09 12:23:48:

Call state for call object 13: connected



[8] 2012/07/09 12:23:48:

call port 24: state code from 100 to 200



[8] 2012/07/09 12:23:48:

call port 23: state code from 183 to 200



[5] 2012/07/09 12:23:48:

SIP Tx tls:95.128.80.120:54209:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120
To: <sip:0325520053@95.128.80.120>;tag=096ec469b7
Call-ID: 1200926091972012122345@95.128.80.120
CSeq: 1 INVITE
Contact: <sip:99999@192.168.1.201:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1705764368 1705764368 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 58306 RTP/AVP 0 8 98 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[7] 2012/07/09 12:23:48:

1200926091972012122345@95.128.80.120: RTP pass-through mode



[7] 2012/07/09 12:23:48:

c580351d@pbx: RTP pass-through mode



[5] 2012/07/09 12:23:48:

SIP Rx tls:95.128.80.120:54209:



ACK sip:99999@192.168.1.201:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac199010494;alias
Max-Forwards: 10
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120
To: <sip:0325520053@95.128.80.120>;tag=096ec469b7
Call-ID: 1200926091972012122345@95.128.80.120
CSeq: 1 ACK
Contact: <sip:96956110529@95.128.80.120:5067;transport=tls>
User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001
Content-Length: 0

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laurent Newbie

laurent

Posted
  • Author
Posted

I try it but it's not ok.

 

and phone are in the same LAN that the Snomone , on Internet they is only the M1000 and he is configured to act as a border controller to solve NAT issue.

for incoming calls the voice is ok just it seems that the SRTP is not correctly negotiated by the snomone. (see my second post)

 

Laurent

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Vodia PBX Grand Master

Vodia PBX

Posted
Posted

and for Incoming calls,

as you can see in the log bellow the M1000 propose the SRTP option in the INVITE SDP but in the 200 OK sendback from the Snomone PBX to the M1000 they is no SRTP proposition.

(for incoming calls RTP is working in both way)

 

any idea how to force snomone to use SRTP ?

 

The M1000 proposes several m-lines in the SDP; the PBX happily picks the first one which does not not have SRTP. If you want that the PBX does SRTP, the first m-line should have the crypto attribute in it (AVP or SAVP does not really matter to the PBX).

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laurent Newbie

laurent

Posted
  • Author
Posted

so for incoming call I have changed the config to only have SDP with SRTP

and now I have exactly the same error that for outgoing calls, SnomOne drop SRTP packet comming from the M1000 and indicate this error:

 

[4] 2012/07/12 11:35:36:Dropped 10 SRTP packets with wrong MAC

[4] 2012/07/12 11:35:38:Dropped 100 SRTP packets with wrong MAC

 

[4] 2012/07/12 11:35:56: Dropped 1000 SRTP packets with wrong MAC

 

regarding the MAC , I see that it's Message Authentication Code used in SRTP, how can we debug this ?

 

Laurent

 

 

[9] 2012/07/12 11:35:35:

Last message repeated 8 times



[8] 2012/07/12 11:35:35:

Received SIP connection 3 from 95.128.80.120:36212



[9] 2012/07/12 11:35:35:

SIP 95.128.80.120:36212: process_client_hello(03014ffea8763d866d889fa9269add497ee328fe4ab0ad93b4aea2cab69b6e8673e600003400c10066006500640063006200390038003500330032002f001600150014001300120011000a00090008000600050004000300ff0100000400230000)



[9] 2012/07/12 11:35:35:

SIP 95.128.80.120:36212: [a50e8e58] send_server_hello(03014ffe9a67b20f9479c4b77ce12741f0ccca7e481e33721501bb534ef58fed8d3100000500)



[9] 2012/07/12 11:35:35:

SIP 95.128.80.120:36212: [a50e8e58] send_certificate(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)



[9] 2012/07/12 11:35:35:

SIP 95.128.80.120:36212: [a50e8e58] send_hello_done()



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] process_client_key_exchange(008013dfeb8f63a8ab71745cf80e252a42893ceae33bef26c0a64a71ce4930b5c545fb965f8df941f722f55b74f6a01475976b0f60767075fda7761c6e4666ecb8ba46f63c16acd6d47d560fe823ea444a7ae32abb80607e532c05e38f722c71d23e2aa9d03ddb36ca70aef46dc63271fa99aed9ca293a1606be79ab52f039d4d09a)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Pre Master Secret(02e32caa4201d1103ef913c75b1f062a58196ba558be96ec7ded891f4ebbd93c566ed9b24bcbcca6aeea3a60ad17f2d46a61b735e8dc3ca7eac23420fa9c5b5431405968e8069ba4d55891b52ef9000301dc3f8d82fb4aecfd353740ea8519fbcede143df6100ba1f3671f7505ba3f0f4ffc114001b8b3cbc22bcd453460c2)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Client Random(4ffea8763d866d889fa9269add497ee328fe4ab0ad93b4aea2cab69b6e8673e6)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Server Random(4ffe9a67b20f9479c4b77ce12741f0ccca7e481e33721501bb534ef58fed8d31)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Client Random(4ffea8763d866d889fa9269add497ee328fe4ab0ad93b4aea2cab69b6e8673e6)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Server Random(4ffe9a67b20f9479c4b77ce12741f0ccca7e481e33721501bb534ef58fed8d31)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Master Secret(3b709cddb1dfe1a570d69534507830742fc3d7c3f501241c0875a56d9ad10b556d0f469be78b59520c5088a3f549c200)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] process_change_cipher_spec(01)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] send_change_cipher_spec(01)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] perform_change_cipher_spec(0005)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Key Block(ff0c51b54c9eb7a9dddca2da5368614456a5f5c7fb794993382ebf035d0b2bb86c785111a2156ab851feaa74c1421f1238ea08f188aa87b155d71e91dabf7d236a24ddf54c526a8d)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Client Write MAC Secret(ff0c51b54c9eb7a9dddca2da5368614456a5f5c7)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Server Write MAC Secret(fb794993382ebf035d0b2bb86c785111a2156ab8)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Client Write Key(51feaa74c1421f1238ea08f188aa87b1)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] Server Write Key(55d71e91dabf7d236a24ddf54c526a8d)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] process_finished(9edac49384bf73b1fb34e154)



[9] 2012/07/12 11:35:36:

SIP 95.128.80.120:36212: [a50e8e58] send_finished(f7a0318d6366144cbb5a348e)



[5] 2012/07/12 11:35:36:

SIP Rx tls:95.128.80.120:36212:



INVITE sip:96956110529@95.128.80.120 SIP/2.0
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias
Max-Forwards: 10
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 1 INVITE
Contact: <sip:96956110529@95.128.80.120:5067;transport=tls>
Allow: ACK,CANCEL,BYE,INFO
User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001
Content-Type: application/sdp
Content-Length: 531
x-changeuri: 1

v=0
o=Dialogic_SDP 1658080552 1658080517 IN IP4 95.128.80.120
s=Dialogic-SIP
c=IN IP4 95.128.80.120
t=0 0
m=audio 8160 RTP/SAVP 8 0 98 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:98 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XiksbNTW1XnGA5ZaqbacEUR5aWwY6Nd9G+wNukYT|2^31|58:1
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0dPFOIy0LfD62d7JP/MMLlOhw23LED2M61Sd/ibD|2^31|220:1




[8] 2012/07/12 11:35:36:

Allocating for call port 0, SIP call id 16585798691272012113534@95.128.80.120 



[9] 2012/07/12 11:35:36:

UDP(IPv4): Opening socket on 0.0.0.0:50160



[9] 2012/07/12 11:35:36:

UDP(IPv4): Opening socket on 0.0.0.0:50161



[9] 2012/07/12 11:35:36:

UDP(IPv6): Opening socket on [::]:50160



[9] 2012/07/12 11:35:36:

UDP(IPv6): Opening socket on [::]:50161



[5] 2012/07/12 11:35:36:

Identify trunk (IP address and domain match) 2



[5] 2012/07/12 11:35:36:

SIP Tx tls:95.128.80.120:36212:



SIP/2.0 100 Trying
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>;tag=3f880a4652
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 1 INVITE
Content-Length: 0





[6] 2012/07/12 11:35:36:

Call-leg 0: Sending RTP for 16585798691272012113534@95.128.80.120 to 95.128.80.120:8160, codec not set yet



[8] 2012/07/12 11:35:36:

Incoming call: Request URI sip:96956110529@95.128.80.120, To is <sip:0325520053@95.128.80.120>



[8] 2012/07/12 11:35:36:

Call from a trunk 2



[8] 2012/07/12 11:35:36:

Trunk Peoplefone@pbx.company.com has country code not set, area code not set



[9] 2012/07/12 11:35:36:

Incoming: formatted From is = "0763770377" <sip:+0763770377@95.128.80.91;user=phone>



[9] 2012/07/12 11:35:36:

Incoming: formatted To is = <sip:0325520053@95.128.80.120;user=phone>



[9] 2012/07/12 11:35:36:

Incoming: formatted URI is = sip:96956110529@pbx.company.com;user=phone



[8] 2012/07/12 11:35:36:

To is <sip:0325520053@95.128.80.120;user=phone>, user 0, domain 1



[8] 2012/07/12 11:35:36:

Send call to extension ERE returned 40



[5] 2012/07/12 11:35:36:

Domain trunk Peoplefone@pbx.company.com sends call to 40 in domain pbx.company.com



[8] 2012/07/12 11:35:36:

Set the To domain based on To user 40@pbx.company.com



[8] 2012/07/12 11:35:36:

Call state for call object 1: idle



[7] 2012/07/12 11:35:36:

Call port 0: set_codecs for 16585798691272012113534@95.128.80.120 codecs "", codec_preference count 6



[8] 2012/07/12 11:35:36:

Call state for call object 1: alerting



[8] 2012/07/12 11:35:36:

Play audio_moh/noise.wav, caching true



[8] 2012/07/12 11:35:36:

Allocating for call port 1, SIP call id 451bca7c@pbx 



[9] 2012/07/12 11:35:36:

UDP(IPv4): Opening socket on 0.0.0.0:60638



[9] 2012/07/12 11:35:36:

UDP(IPv4): Opening socket on 0.0.0.0:60639



[9] 2012/07/12 11:35:36:

UDP(IPv6): Opening socket on [::]:60638



[9] 2012/07/12 11:35:36:

UDP(IPv6): Opening socket on [::]:60639



[7] 2012/07/12 11:35:36:

Call port 1: set_codecs for 451bca7c@pbx codecs "", codec_preference count 6



[9] 2012/07/12 11:35:36:

Using outbound proxy sip:192.168.1.41:4901;transport=tls because of flow-label



[8] 2012/07/12 11:35:36:

call port 1: state code from 0 to 100



[9] 2012/07/12 11:35:36:

Call port 1: update_codecs for 451bca7c@pbx: adding codec pcmu/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 1: update_codecs for 451bca7c@pbx: adding codec pcma/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 1: update_codecs for 451bca7c@pbx: adding codec g722/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 1: update_codecs for 451bca7c@pbx: adding codec g726-32/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 1: update_codecs for 451bca7c@pbx: adding codec gsm/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 1: update_codecs for 451bca7c@pbx: codec_preference size 6, available codecs size 6



[5] 2012/07/12 11:35:36:

SIP Tx tls:192.168.1.41:4901:



INVITE sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>
Call-ID: 451bca7c@pbx
CSeq: 7946 INVITE
Max-Forwards: 70
Contact: <sip:40@192.168.1.201:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Alert-Info: <http://127.0.0.1/Bellcore-dr3>
Content-Type: application/sdp
Content-Length: 421

v=0
o=- 759227115 759227115 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 60638 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8Gi8jkVgW8DzeqpYxFJSOEeTo+NNGp7kJAW/A/mQ
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[8] 2012/07/12 11:35:36:

call port 0: state code from 0 to 100



[9] 2012/07/12 11:35:36:

Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: adding codec pcmu/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: adding codec pcma/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: Other side does not support codec g722/8000



[9] 2012/07/12 11:35:36:

Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: adding codec g726-32/8000 to available list



[9] 2012/07/12 11:35:36:

Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: Other side does not support codec gsm/8000



[9] 2012/07/12 11:35:36:

Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: codec_preference size 6, available codecs size 4



[5] 2012/07/12 11:35:36:

SIP Rx tls:192.168.1.41:4901:



SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
Call-ID: 451bca7c@pbx
CSeq: 7946 INVITE
Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1
Content-Length: 0





[9] 2012/07/12 11:35:36:

Message repetition, packet dropped



[5] 2012/07/12 11:35:36:

SIP Rx tls:192.168.1.41:4901:



SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
Call-ID: 451bca7c@pbx
CSeq: 7946 INVITE
Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1
Require: 100rel
RSeq: 1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0





[5] 2012/07/12 11:35:36:

SIP Tx tls:192.168.1.41:4901:



PRACK sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-f12868f813612b8534cc8634c840c740;rport
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
Call-ID: 451bca7c@pbx
CSeq: 7947 PRACK
Max-Forwards: 70
Contact: <sip:40@192.168.1.201:5061;transport=tls>
RAck: 1 7946 INVITE
Content-Length: 0





[8] 2012/07/12 11:35:36:

Play audio_en/ringback.wav, caching true



[8] 2012/07/12 11:35:36:

call port 0: state code from 100 to 183



[6] 2012/07/12 11:35:36:

Call-leg 0: Codec pcmu/8000 is chosen for call id 16585798691272012113534@95.128.80.120



[5] 2012/07/12 11:35:36:

set codec: codec pcmu/8000 is set to call-leg 0



[5] 2012/07/12 11:35:36:

SIP Tx tls:95.128.80.120:36212:



SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>;tag=3f880a4652
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 1 INVITE
Contact: <sip:99999@192.168.1.201:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Type: application/sdp
Content-Length: 379

v=0
o=- 1200740305 1200740305 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 50160 RTP/SAVP 0 8 98 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Kr2UKk1H96d5DB/dk6xD1Yz2DPyX0Z7do817kx/q
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[5] 2012/07/12 11:35:36:

SIP Rx tls:192.168.1.41:4901:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-f12868f813612b8534cc8634c840c740;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
Call-ID: 451bca7c@pbx
CSeq: 7947 PRACK
Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1
Content-Length: 0





[7] 2012/07/12 11:35:36:

Call 451bca7c@pbx: Clear last request



[4] 2012/07/12 11:35:36:

Dropped 10 SRTP packets with wrong MAC



[4] 2012/07/12 11:35:38:

Dropped 100 SRTP packets with wrong MAC



[5] 2012/07/12 11:35:38:

SIP Rx tls:192.168.1.41:4901:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport=5061
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
Call-ID: 451bca7c@pbx
CSeq: 7946 INVITE
Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1
User-Agent: snom821/8.4.35
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 439

v=0
o=root 988953686 988953687 IN IP4 192.168.1.41
s=call
c=IN IP4 192.168.1.41
t=0 0
m=audio 58908 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JiC3R+xWuA39wKgFmj6ISEdD6jCZrW5I0ZKYOhum
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv




[7] 2012/07/12 11:35:38:

Call 451bca7c@pbx: Clear last INVITE



[6] 2012/07/12 11:35:38:

Call-leg 1: Codec pcmu/8000 is chosen for call id 451bca7c@pbx



[6] 2012/07/12 11:35:38:

Call-leg 1: Sending RTP for 451bca7c@pbx to 192.168.1.41:58908, codec pcmu/8000



[5] 2012/07/12 11:35:38:

set codec: codec pcmu/8000 is set to call-leg 1



[5] 2012/07/12 11:35:38:

SIP Tx tls:192.168.1.41:4901:



ACK sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-e2f6ec7271b167a4dba2f36d1e619818;rport
From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
Call-ID: 451bca7c@pbx
CSeq: 7946 ACK
Max-Forwards: 70
Contact: <sip:40@192.168.1.201:5061;transport=tls>
Content-Length: 0





[7] 2012/07/12 11:35:38:

Determine pass-through mode after receiving response



[8] 2012/07/12 11:35:38:

Call state for call object 1: connected



[8] 2012/07/12 11:35:38:

call port 1: state code from 100 to 200



[8] 2012/07/12 11:35:38:

call port 0: state code from 183 to 200



[5] 2012/07/12 11:35:38:

SIP Tx tls:95.128.80.120:36212:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>;tag=3f880a4652
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 1 INVITE
Contact: <sip:99999@192.168.1.201:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Type: application/sdp
Content-Length: 379

v=0
o=- 1200740305 1200740305 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 50160 RTP/SAVP 0 8 98 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Kr2UKk1H96d5DB/dk6xD1Yz2DPyX0Z7do817kx/q
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[7] 2012/07/12 11:35:38:

16585798691272012113534@95.128.80.120: RTP pass-through mode



[7] 2012/07/12 11:35:38:

451bca7c@pbx: RTP pass-through mode



[5] 2012/07/12 11:35:39:

SIP Rx tls:95.128.80.120:36212:



ACK sip:99999@192.168.1.201:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1185360173;alias
Max-Forwards: 10
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>;tag=3f880a4652
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 1 ACK
Contact: <sip:96956110529@95.128.80.120:5067;transport=tls>
User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001
Content-Length: 0





[8] 2012/07/12 11:35:50:

Packet authenticated by transport layer



[5] 2012/07/12 11:35:52:

SIP Rx tls:192.168.1.41:4901:



BYE sip:40@192.168.1.201:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.41:4901;branch=z9hG4bK-hf8mg4lqk05q;rport
From: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
To: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
Call-ID: 451bca7c@pbx
CSeq: 1 BYE
Max-Forwards: 70
Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1
User-Agent: snom821/8.4.35
RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=652,Tx_Pkts=652,Remote_Tx_Pkts=0
Proxy-Require: buttons
Content-Length: 0





[8] 2012/07/12 11:35:52:

Packet authenticated by transport layer



[5] 2012/07/12 11:35:52:

SIP Tx tls:192.168.1.41:4901:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.1.41:4901;branch=z9hG4bK-hf8mg4lqk05q;rport=4901
From: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey
To: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071
Call-ID: 451bca7c@pbx
CSeq: 1 BYE
Contact: <sip:40@192.168.1.201:5061;transport=tls>
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Length: 0





[7] 2012/07/12 11:35:52:

16585798691272012113534@95.128.80.120: Media-aware pass-through mode



[8] 2012/07/12 11:35:52:

Clearing call port 1, SIP call id 451bca7c@pbx



[8] 2012/07/12 11:35:52:

call port 0: state code from 200 to 486



[5] 2012/07/12 11:35:52:

SIP Tx tls:95.128.80.120:36212:



BYE sip:96956110529@95.128.80.120:5067;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-33dee76e478a1d151aa2114665c1c711;rport
From: <sip:0325520053@95.128.80.120>;tag=3f880a4652
To: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 4306 BYE
Max-Forwards: 70
Contact: <sip:99999@192.168.1.201:5061;transport=tls>
Content-Length: 0





[8] 2012/07/12 11:35:52:

Remove leg 2: call port 1, SIP call id 451bca7c@pbx



[8] 2012/07/12 11:35:52:

Hangup: Call 1 not found



[8] 2012/07/12 11:35:52:

Last message repeated 2 times



[5] 2012/07/12 11:35:52:

SIP 95.128.80.120:36212: Alert(2, 20)



[4] 2012/07/12 11:35:56:

Dropped 1000 SRTP packets with wrong MAC



[5] 2012/07/12 11:35:58:

SIP Rx tls:95.128.80.120:36212:



BYE sip:99999@192.168.1.201:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac2125828410;alias
Max-Forwards: 10
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>;tag=3f880a4652
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 2 BYE
User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001
Content-Length: 0





[5] 2012/07/12 11:35:58:

SIP Tx tls:95.128.80.120:36212:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac2125828410;alias
From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
To: <sip:0325520053@95.128.80.120>;tag=3f880a4652
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 2 BYE
Contact: <sip:99999@192.168.1.201:5061;transport=tls>
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Length: 0





[8] 2012/07/12 11:35:58:

Clearing call port 0, SIP call id 16585798691272012113534@95.128.80.120



[8] 2012/07/12 11:35:58:

Remove leg 1: call port 0, SIP call id 16585798691272012113534@95.128.80.120



[8] 2012/07/12 11:35:58:

Hangup: Call 0 not found



[5] 2012/07/12 11:35:58:

SIP Rx tls:95.128.80.120:36212:



SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.201;branch=z9hG4bK-33dee76e478a1d151aa2114665c1c711;received=62.12.196.104;rport=5061
From: <sip:0325520053@95.128.80.120>;tag=3f880a4652
To: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446
Call-ID: 16585798691272012113534@95.128.80.120
CSeq: 4306 BYE
Contact: <sip:96956110529@95.128.80.120:5067;transport=tls>
Server: Mediant 1000 - MSBG/v.6.60A.011.001
Content-Length: 0

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Vodia PBX Grand Master

Vodia PBX

Posted
Posted

Looks like the problem is in the MKI area (master key index): a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pmP4qAsGAw1ojK/EYz6yU076pauMb79BPc95T+iw|2^31|3:1. The PBX simply does not support this at this time. Can you check if there is a workaround possible on the gateway? If not, we'll have to add this to the PBX, but this will take some time and some testing.

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