Some devices have robotic sound

I have an VTO2000A intercom which I managed to get working.
But there’s a problem.

Whenever I’m in a call with the intercom over Bria or Microsip it works GREAT, it’s just awesome.

The moment I call it from my Gigaset C610 IP I don’t understand anything on the Gigaset side. It’s a completely robotic sound and I can’t understand anything people are saying to the intercom.

On the intercom side I can hear everything fine. People can hear me clearly.

I recorded the call in Freepbx and listened to it to see what the system hears. The recording was smooth as butter, no robotic sounds in the recording.
So something is wrong on the Gigaset side but I cant find anything out from their settings.

Any ideas?

Now I even tried forwarding the call from my Gigaset to a different extension and even to an external mobile phone number and it worked great. Just with these Gigaset handsets I’m having issues. For everything else these Gigasets worked fine, just now they’re causing issues with this intercom.

SIP Debug the connection and watch for the codec settings. This sounds like a transcoding error, but there shouldn’t be any transcoding unless one of your devlces it set up using a different codec than everything else.

On a limb, I’m going to guess your Gigasets are using a different codec than everything else and you’ve never noticed it because the transcoding is working for the rest of the system. Check your codec settings on the Intercom and see what it says.

ALaw to Ulaw to Alaw (for example) could sound pretty crappy.

What I’ve found out during the call is this:

The intercom:

Peer                User/ANR             Call ID                    Format     Hold        Last Message          Peer
192.168.0.166    8001             61d0e4b90e41272             (ulaw|h264)      No       Tx: ACK                    8001

My Bria on iPhone(Works great):

Peer                User/ANR             Call ID                    Format     Hold        Last Message          Peer
192.168.0.1                9                OGE0NzVlNGNlOTR        (ulaw)           No       Rx: ACK                    9

Gigaset Horrible sound:

Peer                User/ANR             Call ID                    Format     Hold        Last Message          Peer
192.168.0.101          1                [email protected]_             (ulaw)           No       Rx: ACK                    1

From what I can tell, they’re all using ulaw.

And I have settup the Gigaset to only use G.711 ulaw and G.711 alaw.

What do you think? :\

Pick one - ULaw for “'Murica” and ALaw for Anything Else.

The Intercom will accept H264 or ULaw. SIP DEBUG will give you more insight into what’s actially happening on the connection. If it’s actually using ULaw everywhere, I think this might be a wrong rabbit hole.

I’m stumped.

This one is from Gigaset

     <--- Transmitting (no NAT) to 192.168.0.101:5060 --->
        SIP/2.0 180 Ringing
        Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
        From: "1" <sip:[email protected]>;tag=2907198688
        To: <sip:[email protected];user=phone>;tag=as50fff393
        Call-ID: [email protected]_168_0_101
        CSeq: 3 INVITE
        Server: FPBX-12.0.76.4(11.21.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Contact: <sip:[email protected]:5060>
        Remote-Party-ID: "Dahua" <sip:[email protected]>;party=called;privacy=off;screen=no
        Content-Length: 0


        <------------>
            -- Connected line update to SIP/1-0000004a prevented.
            -- SIP/8001-0000004b is ringing

        <--- Transmitting (no NAT) to 192.168.0.101:5060 --->
        SIP/2.0 180 Ringing
        Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
        From: "1" <sip:[email protected]>;tag=2907198688
        To: <sip:[email protected];user=phone>;tag=as50fff393
        Call-ID: [email protected]_168_0_101
        CSeq: 3 INVITE
        Server: FPBX-12.0.76.4(11.21.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Contact: <sip:[email protected]:5060>
        Content-Length: 0


        <------------>
            -- Connected line update to SIP/1-0000004a prevented.
            -- SIP/8001-0000004b answered SIP/1-0000004a
        Audio is at 15844
        Adding codec 100003 (ulaw) to SDP
        Adding non-codec 0x1 (telephone-event) to SDP

        <--- Reliably Transmitting (no NAT) to 192.168.0.101:5060 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
        From: "1" <sip:[email protected]>;tag=2907198688
        To: <sip:[email protected];user=phone>;tag=as50fff393
        Call-ID: [email protected]_168_0_101
        CSeq: 3 INVITE
        Server: FPBX-12.0.76.4(11.21.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Contact: <sip:[email protected]:5060>
        Remote-Party-ID: "Dahua" <sip:[email protected]>;party=called;privacy=off;screen=no
        Content-Type: application/sdp
        Content-Length: 234

        v=0
        o=root 49481047 49481047 IN IP4 192.168.0.154
        s=Asterisk PBX 11.21.0
        c=IN IP4 192.168.0.154
        t=0 0
        m=audio 15844 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

        <------------>
               > 0x7f10f4289d20 -- Probation passed - setting RTP source address to 192.168.0.101:5018
        Retransmitting #1 (no NAT) to 192.168.0.101:5060:
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
        From: "1" <sip:[email protected]>;tag=2907198688
        To: <sip:[email protected];user=phone>;tag=as50fff393
        Call-ID: [email protected]_168_0_101
        CSeq: 3 INVITE
        Server: FPBX-12.0.76.4(11.21.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Contact: <sip:[email protected]:5060>
        Remote-Party-ID: "Dahua" <sip:[email protected]>;party=called;privacy=off;screen=no
        Content-Type: application/sdp
        Content-Length: 234

        v=0
        o=root 49481047 49481047 IN IP4 192.168.0.154
        s=Asterisk PBX 11.21.0
        c=IN IP4 192.168.0.154
        t=0 0
        m=audio 15844 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

        ---

        <--- SIP read from UDP:192.168.0.101:5060 --->
        ACK sip:[email protected]:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKfd7b6556238dac9dec0f203567b39ce6;rport
        From: "1" <sip:[email protected]>;tag=2907198688
        To: <sip:[email protected];user=phone>;tag=as50fff393
        Call-ID: [email protected]_168_0_101
        CSeq: 3 ACK
        Contact: <sip:[email protected]:5060>
        Authorization: Digest username="1", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="6ade81af", response="09f9da7ec817abb78eb9235c39303c0e"
        Max-Forwards: 70
        User-Agent: C610 IP/42.076.00.000.000
        Content-Length: 0

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

        <--- SIP read from UDP:192.168.0.101:5060 --->
        ACK sip:[email protected]:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKfd3f422bd105d125d315074885afca59;rport
        From: "1" <sip:[email protected]>;tag=2907198688
        To: <sip:[email protected];user=phone>;tag=as50fff393
        Call-ID: [email protected]_168_0_101
        CSeq: 3 ACK
        Contact: <sip:[email protected]:5060>
        Authorization: Digest username="1", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="6ade81af", response="09f9da7ec817abb78eb9235c39303c0e"
        Max-Forwards: 70
        User-Agent: C610 IP/42.076.00.000.000
        Content-Length: 0

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



This one is from the intercom debug:


<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Connected line update to SIP/8001-00000050 prevented.
    -- SIP/1-00000051 is ringing

<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Connected line update to SIP/8001-00000050 prevented.
    -- SIP/1-00000051 answered SIP/8001-00000050
Audio is at 15324
Adding codec 100003 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1821112262 1821112262 IN IP4 192.168.0.154
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.0.154
t=0 0
m=video 0 RTP/AVP 96
m=audio 15324 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.0.166:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK929229632
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x21a8d10 -- Probation passed - setting RTP source address to 192.168.0.101:5004
       > 0x7f10f4279f70 -- Probation passed - setting RTP source address to 192.168.0.166:20000

<--- SIP read from UDP:192.168.0.166:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK2010863511
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="8001", realm="asterisk", nonce="2e87e257", uri="sip:[email protected]:5060", response="55cda55e97e16028a57f1e1f8ee572e2", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.166:5060 (no NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK2010863511;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

This one is from intercom

<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Connected line update to SIP/8001-00000050 prevented.
    -- SIP/1-00000051 is ringing

<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Connected line update to SIP/8001-00000050 prevented.
    -- SIP/1-00000051 answered SIP/8001-00000050
Audio is at 15324
Adding codec 100003 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1821112262 1821112262 IN IP4 192.168.0.154
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.0.154
t=0 0
m=video 0 RTP/AVP 96
m=audio 15324 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.0.166:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK929229632
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x21a8d10 -- Probation passed - setting RTP source address to 192.168.0.101:5004
       > 0x7f10f4279f70 -- Probation passed - setting RTP source address to 192.168.0.166:20000

<--- SIP read from UDP:192.168.0.166:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK2010863511
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="8001", realm="asterisk", nonce="2e87e257", uri="sip:[email protected]:5060", response="55cda55e97e16028a57f1e1f8ee572e2", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.166:5060 (no NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK2010863511;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Well, it looks like the both are indeed using uLaw (PCMU 8000)

Is there an firmware update for the device…

Also, in your sound tests, it wasn’t clear if the issue is in one direction or both. Can you hear ok on the other end when talking into the intercom?

Hi!
The Gigaset is on the latest firmware.

The sound test I did had these results:

You could hear everything fine on the intercom side
The recorded sound on freepbx (Call recording) also provided information that the recorded sounds coming from the intercom microphone were fine.
On the Gigaset phone you could not understand anything the person said through the intercom.

I really don’t know.

One direction with one phone manufacturer doesn’t work, but the sound recording from the device in the same test work.

This tells me we can stop looking at the Intercom. This has got to be a problem on the Gigaset side. I have no idea what, but the fact that everything works correctly “at Asterisk” and doesn’t at the phone makes it seem pretty clear that your problem is in the phone. The other thing that makes it strange is that all of the phones from that manufacturer are having the problem.

The more you troubleshoot and report, the more it sounds like your phones are the problem.

Ok so here’s a small update.

Today I tried again and for some reason the first couple of seconds it works and then it deteriorates really fast and never recovers. At least it feels like that. I did this after a total reboot of everything in my house (power outage).

Then the next thing I noticed is that if i speak really slowly I can somehow make out what’s being said. Almost sounds like it’s loosing packets.

I don’t know how to explain this, and it’s going to sound funny.

I made a long sound like Aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa and it broke up and sounded like this
Aaaaaaaa aaaaaaaaaaa aaaaaaaa aaaaaaaaa aaaaaaaaaaaaa While also still sounding robotic.

I don’t know how else to put it into writing :smiley: Sounds like a jitter problem?

You can also try using different codecs or grabbing some documentation on your intercom model to track down what is used by default. Could also be a hardware issue. Interference could cause the packets to break up or fail to be received. Is there any way you can attach a picture of your setup, such as connection(s) to and from each system in question?

The problem is that I don’t think the problem is in the Intercom.

The server hears the traffic to and from the intercom just fine, but the one style of phone on the other side of the server loses it. The phone itself works fine in other cases, just not when working with the Intercom.

Lost packets or and IP address conflict could also be issues, so it is something that should be looked at.

I found that sometimes the VTO2000 switches its audio format from ulaw. In my case it changed to pcm s16be. It seems to boot up in this mode and only changes after the first call to the doorphone. I saw this problem using Videolan’s codec information to monitor the rtsp stream. Here is an example with default user into:

rtsp://admin:[email protected]:554/cam/realmonitor?channel=1&subtype=0

These units are really intended for Dahua’s proprietary intercom system and offer very basic SIP operation in Asterisk mode, there are no call progress messages for example. Didn’t bother troubleshooting further as never installed due to this and other issues. The camera works fine though…