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RTP timeout Error

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We are seeing these from a recently installed CS410

The call between sip:123-456-0000@localhost;user=phone and sip:anonymous@localhost;user=phone has been disconnected because of media timeout (120 seconds), 0/5931 packets have been received/sent

 

We are seeing these on all three analog ports...

 

My guess is the call is ended, but the PBX is not detected the hangup and timing out...

 

Investigating this, the first attempt to reach the WEB interface failed, and in a moment or two we recieved the "System start" email from the PBX we just tried to reach.

 

It's as if the HTTP stream forced the reset and this isn't the first time these two events seemed related..

 

Running 3.4.0.3194 (Linux)

MSP 828_v2_03_01 Release 2.4.2

 

The FXS interface is a Arris Touchstone TM504G/NA cable DOCSIS 4 port FXS media Gateway

 

And the installation tech had no details on CPC timing,,, This device is commonly used by Comcast and other Cable providers when selling dial-tone..

Does anyone have the details on the CPC settings for this device?

 

Cheers,

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The FXS interface is a Arris Touchstone TM504G/NA cable DOCSIS 4 port FXS media Gateway

 

The Cable Carrier in question has agreed to get me the answers to our questions regarding how the default settings of the FXS media gateway is provisioned from the switch. What are the specific questions to ask, so that we can program the CS410 for best performance?

 

IE Disconnect Wink Timing?

 

We'll Post what we learn for all to know..

 

Cheers

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The call between sip:123-456-0000@localhost;user=phone and sip:anonymous@localhost;user=phone has been disconnected because of media timeout (120 seconds), 0/5931 packets have been received/sent

 

The default CPC value is 400 ms. This works with most of the cases. Also, you can change value for "Detect polarity change" and try the problem goes away.

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The above is in place.. we are awaiting detail from the cable company on all issues for the settings in the ARRIS 504G media gateway. Hopefully we can delay until we get this information... How do we get to the advanced PSTN interface settings?

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The above is in place.. we are awaiting detail from the cable company on all issues for the settings in the ARRIS 504G media gateway. Hopefully we can delay until we get this information... How do we get to the advanced PSTN interface settings?

 

The PSTN settings are in /etc/sipfxo.conf file. There aren't too many settings though.

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The PSTN settings are in /etc/sipfxo.conf file. There aren't too many settings though.

 

 

In the WIKI / Support site on the CS410 appliance a screen shows the frequency / cadences, is that screen still accessible?

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In the WIKI / Support site on the CS410 appliance a screen shows the frequency / cadences, is that screen still accessible?

 

I believe, those settings(and the screen shots) were part of 2.x and may be early 3.0.x version. We did not update he wiki pages.

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Im getting a similar report:

 

The call between sip:5555551234@cust_domain:5060;user=phone and

sip:900@cust_domain has been disconnected because of media timeout

(3600 seconds), 1036/2178 packets have been received/sent

 

I get a few of these everyday. "900@cust_domain" is the Auto Attendant.

 

I should mention that this customer is using Polycoms (IP331's mostly and PnP supported with a pbxnsip license).

 

Any suggestions on getting these alerts cleaned up? Is this a setting on the Auto Attendant? Maybe its not acknowledging the BYE signal?

 

Is this a compatibility issues with the polycoms? (I have had RTP timeouts on all snom device domains as well, though not nearly as frequent). Should I lower the media timeout to something smaller, like 60 seconds?

 

Thanks in advance.

 

Sudo

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What version are you on? I remember there was an issue when the Polycom muted the call. Then the regular RTP traffic is suspended and the PBX receives only SID packets. I believe that was fixed years ago, thus my question what version you are running.

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What version are you on? I remember there was an issue when the Polycom muted the call. Then the regular RTP traffic is suspended and the PBX receives only SID packets. I believe that was fixed years ago, thus my question what version you are running.

 

Version: 2011-4.5.0.1050 Coma Berenicids (CentOS64)

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In that version the mute problem is definitively fixed. Maybe you can pay attention how many seconds it takes before the call gets disconnected. Also, you can set the flag in the admin part of the PBX to send out an email with the SIP trace in the attachment, so that you get a better idea what is going on without the need to go to PCAP.

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Here is the sip trace:

 

2012/11/26 12:00:36 Tx: tcp:192.168.1.1:13886 (1017 bytes)

INVITE sip:200@10.1.1.1;transport=tcp SIP/2.0

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>

Call-ID: 3236d169@pbx

CSeq: 24571 INVITE

Max-Forwards: 70

Contact: <sip:200@cust_ip_add:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids

Content-Type: application/sdp

Content-Length: 382

 

v=0

o=- 325608696 325608696 IN IP4 cust_ip_add

s=-

c=IN IP4 cust_ip_add

t=0 0

m=audio 56626 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

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

2012/11/26 12:00:36 Rx: tcp:192.168.1.1:13886 (446 bytes)

SIP/2.0 100 Trying

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

CSeq: 24571 INVITE

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Accept-Language: en

Content-Length: 0

 

2012/11/26 12:00:37 Rx: tcp:192.168.1.1:13886 (512 bytes)

SIP/2.0 180 Ringing

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

CSeq: 24571 INVITE

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Allow-Events: talk,hold,conference

Accept-Language: en

Require: 100rel

RSeq: 8193

Content-Length: 0

 

2012/11/26 12:00:37 Tx: tcp:192.168.1.1:13886 (446 bytes)

PRACK sip:200@10.1.1.1;transport=tcp SIP/2.0

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-0bb1d0d9c3f074fe2fbe2e5ccd5a1fbd;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

Call-ID: 3236d169@pbx

CSeq: 24572 PRACK

Max-Forwards: 70

Contact: <sip:200@cust_ip_add:5060;transport=tcp>

RAck: 8193 24571 INVITE

Content-Length: 0

 

2012/11/26 12:00:37 Rx: tcp:192.168.1.1:13886 (441 bytes)

SIP/2.0 200 OK

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-0bb1d0d9c3f074fe2fbe2e5ccd5a1fbd;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

CSeq: 24572 PRACK

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Accept-Language: en

Content-Length: 0

 

2012/11/26 12:00:43 Rx: tcp:192.168.1.1:13886 (766 bytes)

SIP/2.0 200 OK

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

CSeq: 24571 INVITE

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Accept-Language: en

Content-Type: application/sdp

Content-Length: 193

 

v=0

o=- 1353881399 1353881399 IN IP4 10.1.1.1

s=Polycom IP Phone

c=IN IP4 10.1.1.1

t=0 0

m=audio 2236 RTP/AVP 0 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

2012/11/26 12:00:43 Tx: tcp:192.168.1.1:13886 (417 bytes)

ACK sip:200@10.1.1.1;transport=tcp SIP/2.0

Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-a35f384c9690c01159e767ce6f4dcee1;rport

From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

Call-ID: 3236d169@pbx

CSeq: 24571 ACK

Max-Forwards: 70

Contact: <sip:200@cust_ip_add:5060;transport=tcp>

Content-Length: 0

 

2012/11/26 12:01:35 Rx: tcp:192.168.1.1:13886 (864 bytes)

INVITE sip:200@cust_ip_add:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK58b41ef7F5AAB32

From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

CSeq: 1 INVITE

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Accept-Language: en

Supported: 100rel,replaces

Allow-Events: talk,hold,conference

Max-Forwards: 70

Content-Type: application/sdp

Content-Length: 205

 

v=0

o=- 1353881399 1353881400 IN IP4 10.1.1.1

s=Polycom IP Phone

c=IN IP4 10.1.1.1

t=0 0

a=sendonly

m=audio 2236 RTP/AVP 0 101

a=sendonly

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

2012/11/26 12:01:35 Tx: tcp:192.168.1.1:13886 (843 bytes)

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK58b41ef7F5AAB32;rport=13886;received=192.168.1.1

From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

Call-ID: 3236d169@pbx

CSeq: 1 INVITE

Contact: <sip:200@cust_ip_add:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids

Content-Type: application/sdp

Content-Length: 237

 

v=0

o=- 325608696 325608696 IN IP4 cust_ip_add

s=-

c=IN IP4 cust_ip_add

t=0 0

m=audio 56626 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=recvonly

2012/11/26 12:01:35 Rx: tcp:192.168.1.1:13886 (557 bytes)

ACK sip:200@cust_ip_add:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bKd2d05b4862B5D81B

From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

CSeq: 1 ACK

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Accept-Language: en

Max-Forwards: 70

Content-Length: 0

 

2012/11/26 12:01:38 Rx: tcp:192.168.1.1:13886 (568 bytes)

REFER sip:200@cust_ip_add:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK7bb88c1e4E0E8A51

From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

CSeq: 2 REFER

Call-ID: 3236d169@pbx

Contact: <sip:200@10.1.1.1;transport=tcp>

User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439

Accept-Language: en

Refer-To: sip:210@cust_domain.com:5060;user=phone

Referred-By: <sip:200@cust_domain.com>

Max-Forwards: 70

Content-Length: 0

 

2012/11/26 12:01:38 Tx: tcp:192.168.1.1:13886 (431 bytes)

SIP/2.0 202 Accepted

Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK7bb88c1e4E0E8A51;rport=13886;received=192.168.1.1

From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97

To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399

Call-ID: 3236d169@pbx

CSeq: 2 REFER

Contact: <sip:200@cust_ip_add:5060;transport=tcp>

User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids

Content-Length: 0

 

Any Thoughts? I think I remember reading somewhere that Polycoms have issues with the REFER method. Dont know if its the same for PRACK?

 

Thanks again - Sudo

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Yes there were changes in the transfer area. The big question is who disconnects the call after a blind transfer. I guess we need to re-visit the topic with the Polycoms.

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