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--- Doorbird - Audio not routing (http://www1.pbxes.com/forum/threadid.php?threadid=1534763861)

Posted by holbrook on 20.08.2018 at 12:17:

Audio not routing

Still trying to get Doorbird door entry phone to work.

If I configure it to call a local extension directly it works with no problem, video and audio route. If I go via PBXes the video routes but not the audio.

In both cases the phone reports a PMCU audio connection, but through PBXes the audio from Doorbird to phone extension reports 0bps dataa rate.

Any ideas?

Extract from SIP Trace below . Doorbird is holbrook-601 (IP 192.168.8.4) calling holbrook-105 (IP 192.168.8.10cool

Aug 20 11:53:14 VERBOSE[22242] logger.c:
<-- SIP read
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.8.4:5060;rport;branch=z9hG4bKPj2c4a573c-318d-4e16-9315-0d78e4537e3d
Max-Forwards: 70
From: sip:[email protected];tag=1b3da137-865c-4eae-89ad-f1f723856236
P-src-ip: 86.152.199.14
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: fbb14a75-e672-4af1-8965-fab2ad098539
CSeq: 12536 INVITE
Call-Info: <http://192.168.8.4/bha-api/image.cgi> ;purpose=icon, <http://192.168.8.4/bha-api/> ;purpose=info
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Proxy-Authorization: Digest username="holbrook-601", realm="pbxes.org", nonce="1aa70fb81f30e1486a15168c4cbb5470753ffd57", uri="sip:[email protected]", response="6e85558706480f20ddbf6ddae48b2617"
Content-Type: application/sdp
Content-Length: 541
v=0
o=- 3743751201 3743751201 IN IP4 192.168.8.4
s=doorbird
b=AS:84
t=0 0
a=X-nat:0
m=audio 4012 RTP/AVP 0 8 96
c=IN IP4 192.168.8.4
b=TIAS:64000
a=rtcp:4013 IN IP4 192.168.8.4
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
m=video 4014 RTP/AVP 97
c=IN IP4 192.168.8.4
a=rtcp:4015 IN IP4 192.168.8.4
a=sendonly
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1; sprop-parameter-sets=Z00AKNoBQBbpUgAABPAAAGLAwIAB6EgACJVF73wvCIRq,aO48gA==
Aug 20 11:53:14 VERBOSE[22242] logger.c: --- (17 headers 21 lines)Aug 20 11:53:14 VERBOSE[22242] logger.c: --- (17 headers 21 lines)---
Aug 20 11:53:14 VERBOSE[22242] logger.c: Using INVITE request as basis request - fbb14a75-e672-4af1-8965-fab2ad098539
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found user 'holbrook-601'
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP audio format 0
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP audio format 8
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP audio format 96
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP video format 97
Aug 20 11:53:14 VERBOSE[22242] logger.c: Peer audio RTP is at port 192.168.8.4:4012
Aug 20 11:53:14 VERBOSE[22242] logger.c: Peer video RTP is at port 192.168.8.4:4014
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format PCMU
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format PCMA
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format telephone-event
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format H264
Aug 20 11:53:14 VERBOSE[22242] logger.c: Capabilities: us - 0x38161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x200000 (h264), combined - 0x20000c (ulaw|alaw|h264)
Aug 20 11:53:14 VERBOSE[22242] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 20 11:53:14 VERBOSE[22242] logger.c: Looking for 105 in from-internal (domain pbxes.com)
Aug 20 11:53:14 VERBOSE[22242] logger.c: list_route: hop: <sip:[email protected]:5060;ob>
Aug 20 11:53:14 VERBOSE[22242] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.4:5060;branch=z9hG4bKPj2c4a573c-318d-4e16-9315-0d78e4537e3d;received=144.76.38.78;rport=49530
From: sip:[email protected];tag=1b3da137-865c-4eae-89ad-f1f723856236
To: sip:[email protected]
Call-ID: fbb14a75-e672-4af1-8965-fab2ad098539
CSeq: 12536 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]:27756>
Content-Length: 0
---
Aug 20 11:53:15 VERBOSE[8529] logger.c: We're at 144.76.38.78 port 41968
Aug 20 11:53:15 VERBOSE[8529] logger.c: Video is at 144.76.38.78 port 37822
Aug 20 11:53:15 VERBOSE[22242] logger.c: 12 headers, 3 lines
Aug 20 11:53:15 VERBOSE[22242] logger.c: Reliably Transmitting (NAT)
NOTIFY sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 144.76.38.78:27756;branch=z9hG4bK78655d48;rport
From: "Unknown" <sip:[email protected]:27756>;tag=as29e835e6
To: <sip:[email protected]:5060;ob>
Contact: <sip:[email protected]:27756>
Call-ID: [email protected]
CSeq: 102 NOTIFY
User-Agent: PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 87
Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)
---
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x1000 (g722) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x10 (g726) to SDP
Aug 20 11:53:15 VERBOSE[22242] logger.c: Scheduling destruction of call '[email protected]' in 15000 ms
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x400 (ilbc) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x200 (speex) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x2 (gsm) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x100000 (h263p) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x80000 (h263) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x200000 (h264) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: 14 headers, 19 lines
Aug 20 11:53:15 VERBOSE[8529] logger.c: Reliably Transmitting (NAT)
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 144.76.38.78:27756;branch=z9hG4bK09ab8320;rport
From: "Front Door" <sip:[email protected]:27756>;tag=as09401853
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:27756>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: PBX
Max-Forwards: 70
Date: Mon, 20 Aug 2018 10:53:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
P-src-ip: 86.152.199.14
Content-Type: application/sdp
Content-Length: 457
v=0
o=root 12625 12625 IN IP4 144.76.38.78
s=session
c=IN IP4 144.76.38.78
t=0 0
m=audio 41968 RTP/AVP 0 9 8 111 97 110 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 37822 RTP/AVP 103 34 99
a=rtpmap:103 h263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
---
Aug 20 11:53:15 VERBOSE[8529] logger.c: -- Called holbrook-105
Aug 20 11:53:15 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 144.76.38.78:5060;branch=z9hG4bK09ab8320;rport=5060
From: "Front Door" <sip:[email protected]:27756>;tag=as09401853
P-src-ip: 86.152.199.14
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.198
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (11 headers 0 lines)Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (11 headers 0 lines)---
Aug 20 11:53:15 VERBOSE[22242] chan_sip.c: SIP response 100 to standard invite
Aug 20 11:53:15 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 200 OK
Via: SIP/2.0/UDP 144.76.38.78:5060;rport=5060;received=144.76.38.78;branch=z9hG4bK78655d48
Call-ID: [email protected]
From: "Unknown" <sip:[email protected]>;tag=as29e835e6
P-src-ip: 86.152.199.14
To: <sip:[email protected];ob>;tag=z9hG4bK78655d48
CSeq: 102 NOTIFY
Content-Length: 0
Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (8 headers 0 lines)Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (8 headers 0 lines)---
Aug 20 11:53:15 VERBOSE[22242] logger.c: Destroying call '[email protected]'
Aug 20 11:53:15 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 144.76.38.78:5060;branch=z9hG4bK09ab8320;rport=5060
From: "Front Door" <sip:[email protected]:27756>;tag=as09401853
P-src-ip: 86.152.199.14
To: <sip:[email protected]:5060>;tag=2104099985
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.198
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (13 headers 0 lines)Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (13 headers 0 lines)---
Aug 20 11:53:15 VERBOSE[22242] chan_sip.c: SIP response 180 to standard invite
Aug 20 11:53:15 VERBOSE[8529] logger.c: -- SIP/holbrook-105-1ff6 is ringing
Aug 20 11:53:15 VERBOSE[8529] logger.c: Transmitting (NAT)
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.4:5060;branch=z9hG4bKPj2c4a573c-318d-4e16-9315-0d78e4537e3d;received=144.76.38.78;rport=49530
From: sip:[email protected];tag=1b3da137-865c-4eae-89ad-f1f723856236
To: sip:[email protected];tag=as0a404948
Call-ID: fbb14a75-e672-4af1-8965-fab2ad098539
CSeq: 12536 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]:27756>
Content-Length: 0
---
Aug 20 11:53:18 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 200 OK
Via: SIP/2.0/UDP 144.76.38.78:5060;branch=z9hG4bK09ab8320;rport=5060
From: "Front Door" <sip:[email protected]:27756>;tag=as09401853
P-src-ip: 86.152.199.14
To: <sip:[email protected]:5060>;tag=2104099985
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.198
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 410
v=0
o=holbrook-105 8000 8000 IN IP4 192.168.8.108
s=SIP Call
c=IN IP4 192.168.8.108
t=0 0
m=audio 5004 RTP/AVP 0 8 9 101
a=sendrecv
a=rtcp:5005 IN IP4 192.168.8.108
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 103 34 99
a=rtpmap:103 h263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
Aug 20 11:53:18 VERBOSE[22242] logger.c: --- (13 headers 18 lines)Aug 20 11:53:18 VERBOSE[22242] logger.c: --- (13 headers 18 lines)---
Aug 20 11:53:18 VERBOSE[22242] chan_sip.c: SIP response 200 to standard invite
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 0
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 8
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 9
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 101
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP video format 103
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP video format 34
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP video format 99
Aug 20 11:53:18 VERBOSE[22242] logger.c: Peer audio RTP is at port 192.168.8.108:5004
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format PCMU
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format PCMA
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format G722
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format telephone-event
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format h263-1998
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format H263
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format H264
Aug 20 11:53:18 VERBOSE[22242] logger.c: Capabilities: us - 0x38161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p|h264), peer - audio=0x100c (ulaw|alaw|g722)/video=0x380000 (h263|h263p|h264), combined - 0x38100c (ulaw|alaw|g722|h263|h263p|h264)
Aug 20 11:53:18 VERBOSE[22242] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 20 11:53:18 VERBOSE[22242] logger.c: list_route: hop: <sip:[email protected]:5060>
Aug 20 11:53:18 VERBOSE[22242] logger.c: set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
Aug 20 11:53:18 VERBOSE[22242] logger.c: set_destination: set destination to 192.168.8.108, port 5060
Aug 20 11:53:18 VERBOSE[22242] logger.c: Transmitting (NAT)
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 144.76.38.78:27756;branch=z9hG4bK112ecf2d;rport
From: "Front Door" <sip:[email protected]:27756>;tag=as09401853
To: <sip:[email protected]:5060>;tag=2104099985
Contact: <sip:[email protected]:27756>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: PBX
Max-Forwards: 70
Content-Length: 0
---

Posted by i-p-tel on 20.08.2018 at 17:45:

RE: Audio not routing

An RTP trace needs to be done. Please call us when you are ready to make another testcall between the two.

Posted by holbrook on 03.09.2018 at 11:36:

RE: Audio not routing

Hi. Thanks for this offer.

Would either 1pm UK time (2pm DE) today 3rd August or 10am UK (11am DE) tomorrow 4th be convenient?

Posted by i-p-tel on 07.09.2018 at 10:12:

RE: Audio not routing

OK, please make another testcall like the one yesterday. Have already activated another trace on your account.

Posted by holbrook on 07.09.2018 at 10:58:

RE: Audio not routing

Just did a test call (~ 09:58 UTC). Same issue - audio drops out after a second, video fine

Posted by i-p-tel on 10.09.2018 at 07:15:

RE: Audio not routing

It seems the call is switching into a hold state. Please add a music on hold to your account, to shed light on this.

Posted by holbrook on 10.09.2018 at 10:54:

RE: Audio not routing

Hi Pascal. Thanks for continuing investigations.

I added a MoH track and tested again (~09:52 UTC) and no apparent change - I certainly don't hear music at either end

13/9 Any update on this please?

Posted by i-p-tel on 06.10.2018 at 10:29:

RE: Doorbird - Audio not routing

Sorry we missed analyzing your log in time. It gets cleared every month. Please repeat and give me date and time of your testcall.

Posted by holbrook on 08.10.2018 at 10:06:

RE: Doorbird - Audio not routing

Thanks.

Just done another test call. 2018-10-08 at 09.04 UTC

Posted by i-p-tel on 12.10.2018 at 17:47:

RE: Doorbird - Audio not routing

I have activated a SIP trace to catch the re-invite that is coming from your doorbird. It is not included in your initial SIP trace. Make another testcall and again quote date/time, please.

Posted by holbrook on 12.10.2018 at 19:40:

RE: Doorbird - Audio not routing

OK, just done a test call ~ 2018-10-12 18.39UTC

Posted by i-p-tel on 13.10.2018 at 14:32:

RE: Doorbird - Audio not routing

Great. The trace shows the reason for the missing audio connection. Instead of the sendrecv mode in its initial INVITE message the following one puts the call on hold:

s=doorbird
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 96
c=IN IP4 82.13.188.220
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.24.28
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
m=video 4002 RTP/AVP 97
c=IN IP4 82.13.188.220
a=rtcp:4003 IN IP4 192.168.24.28
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1; sprop-parameter-sets=Z00AKNoBQBbpUgAABPAAAGLAwIAB6EgACJVF73wv\
CIRq,aO48gA==
a=sendonly
Oct 12 19:39:24 VERBOSE[22242] logger.c: --- (16 headers 20 lines)Oct 12 19:39:24 VERBOSE[22242] logger.c: --- (16 hea\
ders 20 lines)---
Oct 12 19:39:24 VERBOSE[22242] logger.c: Ignoring this INVITE request
Oct 12 19:39:24 VERBOSE[22242] chan_sip.c: Got a SIP re-invite for call 965d4342-30db-4735-b516-c9e645b4c90c

Notice the sendonly in there. Please ask the manufacturer of your device why it chooses to put the call on hold by sending another INVITE message.

Posted by holbrook on 15.10.2018 at 12:39:

RE: Doorbird - Audio not routing

Thanks for this. I will get back to the manufacturer.

Is there anything we can do in the meantime to ensure this error does not trip up PBXes? When I connect directly to phones they all seem to cope with it.

Posted by i-p-tel on 15.10.2018 at 16:27:

RE: Doorbird - Audio not routing

Thanks for your comment. The manufacturer's implementation is probably correct but likely to be incompatible. We've changed our code. Please try if it works for you now.

Posted by holbrook on 15.10.2018 at 17:55:

RE: Doorbird - Audio not routing

Thank you - that works. Your timing was brilliant, as I'm set to install the unit tomorrow and thought I was going to have to explain it only calling 1 extension directly.

Posted by i-p-tel on 15.10.2018 at 19:21:

RE: Doorbird - Audio not routing

Great and thanks for your patience!

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