[SIPForum-discussion] SIP Error 481

Fri Sep 21 11:10:04 UTC 2012

Hey Bulandus,

In the mean while can you ask the IP-PBX guys and let them know that IP-PBX
is not sending Require:timer even if it supports Session-Expiry mechanism.I
am sure they will be happy to know that you raised this issue.Awaiting for
your reply over the same.

Regards
GS-LAB-PVTLTD,PUNE
Abhisek Acharya

On Fri, Sep 21, 2012 at 4:37 PM, Abhisek Acharya
<abhisek.acharya at gmail.com>wrote:

> Hey
>
> I checkd the trace that you sent and this is a failed call.Call is being
> CANCELLED by the calling party here.
>
> Regards
> Abhisek Acharya
>
>
> On Fri, Sep 21, 2012 at 4:35 PM, Abhisek Acharya <
> abhisek.acharya at gmail.com> wrote:
>
>> 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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