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Large TCP SIP messages seem to be chopped

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Large TCP SIP messages seem to be chopped

Hi,

 

I'm seeing some SIP messages arrving on our server from TCP Polycom phones with short/cut SIP messages, almost as if an internal MTU limit has been reached before transmission.

In general we tend to recommend UA provisioning profiles to use TCP (or tls) to avoid MTU problems, which are increasingly common with video / or opus enabled endpoints.

 

Here's a trace. You'll see the reinvite with the authentication parameters is significantly bigger than the first, and at the same time, the bottom part of the SDP is missing. This is causing our server to ignore the message.

 

Am I doing anything wrong?

 

Thanks

 

Adam

 

T 193.89.189.254:34436 -> 193.203.211.158:6060 [AP]
INVITE sip:03330237000@universe.voip.co.uk;user=phone;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 10.45.201.133;branch=z9hG4bK3b13413920E46DA.
From: "Adam" <sip:sip_nfpmdawsqe@universe.voip.co.uk>;tag=396C3D7F-ACEDA430.
To: <sip:03330237000@universe.voip.co.uk;user=phone>.
CSeq: 1 INVITE.
Call-ID: 877cec431d4f1d62583c545b6617d6e2.
Contact: <sip:sip_nfpmdawsqe@10.45.201.133;transport=tcp>.
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER.
User-Agent: PolycomVVX-VVX_411-UA/5.9.1.0615.
Accept-Language: en.
Supported: replaces,100rel.
Allow-Events: conference,talk,hold.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 486.
.
v=0.
o=- 1550662745 1550662745 IN IP4 10.45.201.133.
s=Polycom IP Phone.
c=IN IP4 10.45.201.133.
t=0 0.
a=sendrecv.
m=audio 2244 RTP/AVP 121 9 102 0 8 127 126.
a=rtpmap:121 opus/48000/2.
a=fmtp:121 minptime=10; maxplaybackrate=16000; maxaveragebitrate=24000; sprop-maxcapturerate=16000; usedtx=0.
a=rtpmap:9 G722/8000.
a=rtpmap:102 G7221/16000.
a=fmtp:102 bitrate=32000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:127 telephone-event/8000.
a=rtpmap:126 telephone-event/48000.


T 193.203.211.158:6060 -> 193.89.189.254:34436 [AP]
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/TCP 10.45.201.133;received=193.89.189.254;rport=34436;branch=z9hG4bK3b13413920E46DA.
From: "Adam" <sip:sip_nfpmdawsqe@universe.voip.co.uk>;tag=396C3D7F-ACEDA430.
To: <sip:03330237000@universe.voip.co.uk;user=phone>;tag=489fcdd584739b7762a492202841b98a.0ed9.
CSeq: 1 INVITE.
Call-ID: 877cec431d4f1d62583c545b6617d6e2.
Proxy-Authenticate: Digest realm="fcplatform.com", nonce="5c6d3c780000762fa276db348e5a5f059778b1be7ee6eb69", qop="auth".
Content-Length: 0.
.


T 193.89.189.254:34436 -> 193.203.211.158:6060 [AP]
ACK sip:03330237000@universe.voip.co.uk;user=phone;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 10.45.201.133;branch=z9hG4bK3b13413920E46DA.
From: "Adam" <sip:sip_nfpmdawsqe@universe.voip.co.uk>;tag=396C3D7F-ACEDA430.
To: <sip:03330237000@universe.voip.co.uk;user=phone>;tag=489fcdd584739b7762a492202841b98a.0ed9.
CSeq: 1 ACK.
Call-ID: 877cec431d4f1d62583c545b6617d6e2.
Contact: <sip:sip_nfpmdawsqe@10.45.201.133;transport=tcp>.
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER.
User-Agent: PolycomVVX-VVX_411-UA/5.9.1.0615.
Accept-Language: en.
Max-Forwards: 70.
Content-Length: 0.
.


T 193.89.189.254:34436 -> 193.203.211.158:6060 [A]
INVITE sip:03330237000@universe.voip.co.uk;user=phone;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP 10.45.201.133;branch=z9hG4bK1bbc8d53ED3AA2E4.
From: "Adam" <sip:sip_nfpmdawsqe@universe.voip.co.uk>;tag=396C3D7F-ACEDA430.
To: <sip:03330237000@universe.voip.co.uk;user=phone>.
CSeq: 2 INVITE.
Call-ID: 877cec431d4f1d62583c545b6617d6e2.
Contact: <sip:sip_nfpmdawsqe@10.45.201.133;transport=tcp>.
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER.
User-Agent: PolycomVVX-VVX_411-UA/5.9.1.0615.
Accept-Language: en.
Supported: replaces,100rel.
Allow-Events: conference,talk,hold.
Proxy-Authorization: Digest username="sip_nfpmdawsqe", realm="fcplatform.com", nonce="5c6d3c780000762fa276db348e5a5f059778b1be7ee6eb69", qop=auth, cnonce="rFOhVQkTzfO7y3k", nc=00000001, uri="sip:03330237000@universe.voip.co.uk;user=phone;transport=tcp", response="69930cebf12ba232924773b998483c27", algorithm=MD5.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 486.
.
v=0.
o=- 1550662745 1550662745 IN IP4 10.45.201.133.
s=Polycom IP Phone.
c=IN IP4 10.45.201.133.
t=0 0.
a=sendrecv.
m=audio 2244 RTP/AVP 121 9 102 0 8 127 126.
a=rtpmap:121 opus/48000/2.
a=fmtp:121 minptime=10; maxplaybackrate=16000; maxaveragebitrate=24000; sprop-maxcapturerate=16000; usedtx=0.
a=rtpmap:9 G722/8000.
a=rtpmap:102 G7221/16000.
a=fmtp:102 bitrate=32000.
a=rt
Message 1 of 6
5 REPLIES 5
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Polycom Employee & Community Manager

Re: Large TCP SIP messages seem to be chopped

Hello adam.crisp@firstcomeurope.com ,

 

welcome to the Polycom Community.

Both the community's Must Read First and the community's FAQ reference the basic minimum information a new or follow up post should contain.

This ensures the questions having to be asked are limited and any new or follow up post contains the right amount of details to ensure any voluntary participant within the community does not spend additional time chasing basic information.

As a reminder the basic information asked for:

 

  • Provide the exact Software Version of your Phone
  • Provide the Phone Model
  • Provide the Call Platform (aka openSIP,Skype for Business Online, Skype for Business on Premise, Lync)
  • Additional Polycom Infrastructure (RPRM,PDMS or BToE)
  • If applicable provide a backup of the phone in question

UC Software 4.0.0 or later via the Web Interface Utilities > Phone Backup & Restore > Phone Backup > Phone Backup. Please rename into .TXT or Zip the file to attach.
Since UC Software 5.9.0 simply provide this via the Web Interface Diagnostics > Download Support Information Package

  • If possible provide a Log and either attach them or use the Code Tag.Consult the Troubleshooting Section found within the FAQ if applicable
  • If possible provide the MAC Address or Serial of the device
  • Provide details for example if the issue is a day 1 issue or only happened after an upgrade or any other relevant details
  • For questions around Support please check here

 

Whilst providing some of these details may not directly impact any possible answer the community can provide, it does enable Polycom to have an overview of the current software used. In addition providing all details at the same time allow us to check logs or look up a potential support partners if an issue needs to come into support. It also enables us to verify the entitlement for using features.


Please ensure you always check the FAQ's and/or utilize the community search before posting any new topics or follow up post’s.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 2 of 6
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Re: Large TCP SIP messages seem to be chopped

Hi,

 

Thank you for your reply.

The call was being sent to an Opensips machine:

 

devices tried:

 

PolycomSoundStationIP-SSIP_5000-UA/4.0.4.2906
PolycomSoundPointIP-SPIP_450-UA/4.0.4.2906
PolycomSoundPointIP-SPIP_450-UA/4.0.14.0987


Phone Model VVX 411
Part Number 3111-48450-001 Rev:A
UC Software Version 5.9.1.0615
Updater Version 5.9.7.12459

I am struggling to attached a .txt or .zip - so here' a link:

 

https://www.dropbox.com/s/wsp5ht2ut2elncc/64167f17d6e2_VVX411_26-02-2019_09-44.pbu.txt?dl=0

Message 3 of 6
Highlighted
Polycom Employee & Community Manager

Re: Large TCP SIP messages seem to be chopped

Hello adam.crisp@firstcomeurope.com ,

 

welcome back to the Polycom Community.


Not knowing if this works with the other phones only and not with the VVX I suggest you open this via the Polycom Sales Engineer that supplied the Phone.

 

We cannot troubleshoot issues via the community.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 4 of 6
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Re: Large TCP SIP messages seem to be chopped

OK. The issues I've seen on all the devices on the list, but interesting out of the box firmware of the VVX was OK - but later versions are affected.

 

I'm sorry for asking about this possible bug on a community forum. At least if somebody else has the same problem they'll know they are not alone. If I ever get round to mentioning it to a polycom engineer and if they have a fix, I'll try to come back and mention it.

Message 5 of 6
Highlighted
Polycom Employee & Community Manager

Re: Large TCP SIP messages seem to be chopped

Hello adam.crisp@firstcomeurope.com ,

 

specifying the relevant version would be helpful to anyone finding this post in the future.

 

A service ticket is required so we can escalate this within Polycom and test this with your setup. This cannot be done via the medium of a community forum.

 

Please share the SR number once opened as nothing can be gained from this post otherwise.


Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 6 of 6