[SIPForum-discussion] SIP Error 481

Fri Sep 21 11:05:40 UTC 2012

Hi Dear Bulandus,

Low session-Expiry values are not dangerous but this can create a lot of
mid-dialog request and responses which will unnecessarily over load the
network.
Session-Expiry values should be greater than 90 seconds.It should not be
less than this.

Significance of Require:timer

1.Let us assume UAC sends supported:timer that means it supports
session-expiry mechanism.If the UAS supports Session-Refresher mechanism
then it MUST send Require:timer to kick in the Session-Refresher
mechanism.It does not matter who is the refresher the UAS MUST send
Require:timer to kick in the Session-Expiry method.
I am yet to check the RFC.Let me check RFC 4028 and will get back to
you.Also I will check your new trace and will get back to you as well.

Regards
Abhisek Acharya
GS-LAB,PVT LTD,
PUNE

On Fri, Sep 21, 2012 at 1:32 PM, BULANDUS, Tristan L. <
tlbulandus at pldt.com.ph> wrote:

> Attachment I have here represents an Alcatel IPPBX with almost the same
> scenario but call does not get cut.****
>
> ** **
>
> For your comments please.****
>
> ** **
>
> Thank you very much.****
>
> * *
>
> *Tristan Bulandus*
>
> *Network Services Assurance*
>
> *Customer Network, Network Engineer*
>
> *0919-332-5174*
>
> * *
>
> *PLDT Futsal Team Captain*
>
> [image: Description: cid:image001.jpg at 01CD8757.999FBEA0] ****
>
> ** **
>
> *From:* BULANDUS, Tristan L.
> *Sent:* Friday, September 21, 2012 3:48 PM
> *To:* 'Abhisek Acharya'
> *Cc:* discussion at sipforum.org
> *Subject:* RE: [SIPForum-discussion] SIP Error 481****
>
> ** **
>
> What can “Require:timer” do?****
>
> ** **
>
> Is it really dangerous to have a low session-expiration timer? ****
>
> ** **
>
> Thank you very much.****
>
> * *
>
> *Tristan Bulandus*
>
> *Network Services Assurance*
>
> *Customer Network, Network Engineer*
>
> *0919-332-5174*
>
> * *
>
> *PLDT Futsal Team Captain*
>
> [image: Description: cid:image001.jpg at 01CD8757.999FBEA0] ****
>
> ** **
>
> *From:* Abhisek Acharya [mailto:abhisek.acharya at gmail.com]
> *Sent:* Friday, September 21, 2012 12:44 PM
>
> *To:* BULANDUS, Tristan L.
> *Cc:* discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] SIP Error 481****
>
> ** **
>
> Hey BULANDUS,
>
> If I am not wrong Your call gets disconnected in every five minutes.If
> that is true then in your system session refresher mechanism is not working
> as per the standard.
>
> I saw a INVITE is coming from 10.249.211.20 with Supported:timer value
> and Session-Expiry time with 300 seconds.And in 200 OK response the 10.251.3.153
> MUST send Require:timer but rather it is sending Supported timer which is
> not correct.First thing what you need to check with the AsteriskIP-PBX is
> why it is not sending Require:timer .May be you can raise a BUG or
> something.
>
> Post that I am seeing a lot of  Session-Refresher requests from 10.251.3.153
> IP travelling till the end.And at times it is not getting the proper
> response and and request is pending on the other side.Because the INVITE is
> not processed and yet another comes.
>
> And Also 10.251.3.153 sending so many INVITES with chaning the sdp body
> without getting the response is also a problem.
> And due to some reason 10.249.211.20 removes the state of the dialog
> which it should not do.
>
> FEW SUGGESTIONS
>
> 1.Check Why 10.251.3.153 not replying with Reuire:timer even if it
> supports Session-refresher mechanism.It should not send Supported:timer
> rather it should send Require:timer value.
>
> 2.Change the SessionExpiry value to a big number such as 1800 seconds or
> something becuase small session-expiry values are dangerous to network as
> it will create a lot of congestion with huge number of message flow.
>
> 3.Calls getting disconnected excatly in 5 minutes because
> Session-Refresher value is 300 seconds.481 response has nothing to do with
> the call disconnection as this happens way after the call set up.
>
> Awaiting for your response over the same.
>
> Regards
> Abhisek Acharya
> GS-LAB
> PUNE
>
>
>
>
>
> Please feel free to ask queries if any or if any something is not clear on
> this.
>
>
>
>
>
>
>
>
> ****
>
> On Fri, Sep 21, 2012 at 6:36 AM, BULANDUS, Tristan L. <
> tlbulandus at pldt.com.ph> wrote:****
>
> Team, ****
>
> Appreciate help on below.****
>
>  ****
>
> 10.251.3.153 - Asterisk IPPBX****
>
> 10.249.211.20 - Acme SBC facing Asterisk****
>
> 10.249.212.32 - Acme SBC facing Huawei Softswitch****
>
> 10.249.200.1 - Huawei Softswitch facing PSTN****
>
>  ****
>
> Thank you very much.****
>
> * *****
>
> *Tristan Bulandus*****
>
> *Network Services Assurance*****
>
> *Customer Network, Network Engineer*****
>
> *0919-332-5174*****
>
> * *****
>
> *PLDT Futsal Team Captain*****
>
> [image: Description: cid:image001.jpg at 01CD8757.999FBEA0] ****
>
>  ****
>
> *From:* Abhisek Acharya [mailto:abhisek.acharya at gmail.com]
> *Sent:* Thursday, September 20, 2012 12:37 PM
> *To:* BULANDUS, Tristan L.
> *Cc:* discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] SIP Error 481****
>
>  ****
>
> Hey Buddy,
>
>
> My initial question to you is open the INVITE message and check the FROM
> and TO header.Also try to check the TO header in specific.There should be
> no TO-TAG present in the TO header.As the Initial INVITE never contains a
> TO-TAG.
>
>
> 481 means call-leg does not exist.You are saying call gets disconnected
> after 5 minutes.Which means the call gets disconnected after the session is
> established.Can you please check if the RE-INVITE contains the same FROM
> and TO TAG or not.
>
> Also Please send me the full wireshark trace and IP addresses.So that I
> can quickly check and get back to you.
>
> Regards
> Abhisek ****
>
> On Wed, Sep 19, 2012 at 2:03 PM, BULANDUS, Tristan L. <
> tlbulandus at pldt.com.ph> wrote:****
>
> Hi Team,****
>
>  ****
>
> We have a network set-up of SIP Peering services, with details below:****
>
>  ****
>
> ASTERISK PBX – SESSION BORDER CONTROLLER (PROXY) – HUAWEI SOFTSWITCH (PSTN
> GATEWAY)****
>
> [image: Description: cid:image001.png at 01CD9683.2CB34240]****
>
>  ****
>
> Calls originating from the PSTN Gateway is able to establish connection to
> Asterisk PBX but the call gets cut after 5 minutes. We experience Error 481
> from (SBC - 10.249.211.20) to (Asterisk – 10.251.3.153).****
>
>  ****
>
> Expert advice please on how to resolve this.****
>
>  ****
>
> Thank you very much.****
>
>
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>  ****
>
> ** **
>
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