Comrex fieldtap and Asterisk

tellierflexus · January 10, 2023, 9:06pm

Hello everyone,
I’m trying to use the Comrex Fieldtap app as an sip client for radio journalist use.
But when trying to call an extension, the app hang up directly (see attached log). The app can however call a sip server provided by comrex for testing. Do you have any idea of what is happening ?

<--- SIP read from UDP:86.XX.XX.XX:5060 --->
INVITE sip:1234@192.168.2.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.JAmQL6bkQ;rport
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>
CSeq: 20 INVITE
Call-ID: cAXXfTQjVn
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 289
Contact: <sip:7001@86.XX.XX.XX;transport=udp>;+sip.instance="<urn:uuid:53a451d1-8fea-496b-a938-7dc733c66b33>"
User-Agent: Comrex.FieldTap_iPhone12.1_iOS15.6.1/3.16.5-33-g367bf180 (belle-sip/1.6.3)

v=0
o=7001 2714 1359 IN IP4 192.168.1.27
s=Talk
c=IN IP4 192.168.1.27
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7256 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (13 headers 11 lines) ---
Sending to 86.XX.XX.XX:5060 (NAT)
Sending to 86.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - cAXXfTQjVn
Found peer '7001' for '7001' from 86.XX.XX.XX:5060

<--- Reliably Transmitting (NAT) to 86.XX.XX.XX:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.JAmQL6bkQ;received=86.XX.XX.XX;rport=5060
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as2dee5132
Call-ID: cAXXfTQjVn
CSeq: 20 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7badf9c9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'cAXXfTQjVn' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:86.XX.XX.XX:5060 --->
ACK sip:1234@192.168.2.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.JAmQL6bkQ;rport
Call-ID: cAXXfTQjVn
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as2dee5132
Contact: <sip:7001@86.XX.XX.XX;transport=udp>;+sip.instance="<urn:uuid:53a451d1-8fea-496b-a938-7dc733c66b33>"
Max-Forwards: 70
CSeq: 20 ACK

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:86.XX.XX.XX:5060 --->
INVITE sip:1234@192.168.2.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.bV1cGt8jd;rport
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>
CSeq: 21 INVITE
Call-ID: cAXXfTQjVn
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 289
Contact: <sip:7001@86.XX.XX.XX;transport=udp>;+sip.instance="<urn:uuid:53a451d1-8fea-496b-a938-7dc733c66b33>"
User-Agent: Comrex.FieldTap_iPhone12.1_iOS15.6.1/3.16.5-33-g367bf180 (belle-sip/1.6.3)
Authorization: Digest realm="asterisk", nonce="7badf9c9", algorithm=MD5, username="7001", uri="sip:1234@192.XX.XX.XX:5060", response="a760310bfafb9c5b6ef6cf4a46fa9fc7"

v=0
o=7001 2714 1359 IN IP4 192.168.1.27
s=Talk
c=IN IP4 192.168.1.27
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7256 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (14 headers 11 lines) ---
Sending to 86.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - cAXXfTQjVn
Found peer '7001' for '7001' from 86.XX.XX.XX:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.27:7256
Looking for 1234 in internal (domain 192.168.2.33)
sip_route_dump: route/path hop: <sip:7001@86.XX.XX.XX;transport=udp>

<--- Transmitting (NAT) to 86.XX.XX.XX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.bV1cGt8jd;received=86.XX.XX.XX;rport=5060
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>
Call-ID: cAXXfTQjVn
CSeq: 21 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:1234@192.XX.XX.XX:5060>
Content-Length: 0


<------------>
Audio is at 17490
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 86.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.bV1cGt8jd;received=86.XX.XX.XX;rport=5060
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as26af83ec
Call-ID: cAXXfTQjVn
CSeq: 21 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:1234@192.XX.XX.XX:5060>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 685003043 685003043 IN IP4 192.XX.XX.XX
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 192.XX.XX.XX
t=0 0
m=audio 17490 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:86.XX.XX.XX:5060 --->
BYE sip:1234@192.168.2.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.BEAq39vvv;rport
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as26af83ec
CSeq: 22 BYE
Call-ID: cAXXfTQjVn
Max-Forwards: 70
User-Agent: Comrex.FieldTap_iPhone12.1_iOS15.6.1/3.16.5-33-g367bf180 (belle-sip/1.6.3)
Authorization: Digest realm="asterisk", nonce="7badf9c9", algorithm=MD5, username="7001", uri="sip:1234@192.XX.XX.XX:5060", response="8db3d7f2356e30313a6504442131fcb7"

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

<--- Reliably Transmitting (NAT) to 86.XX.XX.XX:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.bV1cGt8jd;received=86.XX.XX.XX;rport=5060
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as26af83ec
Call-ID: cAXXfTQjVn
CSeq: 21 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<------------>
Sending to 86.XX.XX.XX:5060 (NAT)
Scheduling destruction of SIP dialog 'cAXXfTQjVn' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 86.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK.BEAq39vvv;received=86.XX.XX.XX;rport=5060
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as26af83ec
Call-ID: cAXXfTQjVn
CSeq: 22 BYE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


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

<--- SIP read from UDP:86.XX.XX.XX:5060 --->
ACK sip:1234@192.168.2.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;rport;branch=z9hG4bK.XLxFQUjqQ
From: "ComrexApp" <sip:7001@voip.myassociation.ch>;tag=wjEAUal9-
To: "1234" <sip:1234@192.168.2.33:5060>;tag=as26af83ec
CSeq: 21 ACK
Call-ID: cAXXfTQjVn
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="7badf9c9", algorithm=MD5, username="7001", uri="sip:1234@192.XX.XX.XX:5060", response="a760310bfafb9c5b6ef6cf4a46fa9fc7"
User-Agent: Comrex.FieldTap_iPhone12.1_iOS15.6.1/3.16.5-33-g367bf180 (belle-sip/1.6.3)

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

Have a good day

david551 · January 10, 2023, 9:22pm

You’ve got something on a public address giving Asterisk a media address which is on a private address. Best to fix that at source, but otherwise you will need to enable comedia to work round it.

Also, the request URI domain part has been modified. That suggests you have a rogue SIP ALG gateway. Asterisk will ignore the domain part, but you can expect weird behaviour from any misbehaving ALG.

For anyone else reading this, the obfuscated 192.x.x.x address is NOT a 192.168 one.

tellierflexus · January 10, 2023, 9:29pm

Hello
Huum, i already have comedia activated on this extension

[7001]
type=friend
host=dynamic
secret=***********
context=internal
nat=force_rport,comedia

For the sip alg gateway I have the same problem when using mobile network so don’t think i can change that ?

Hahah sorry for the ip issue again, next time I should ofuscate it another way

system · February 9, 2023, 9:29pm

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