[SIPForum-discussion] SIP Error 481

Abhisek Acharya abhisek.acharya at gmail.com
Fri Sep 21 04:43:43 UTC 2012


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.****
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
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> Post to the list at discussion at sipforum.org****
>
> ** **
>
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