[OpenSIPS-Users] convert 180 to 183 after the fact

[Jeff Pyle]

Muhammad,

That makes sense.  I think in my case I would have to strip the SDP as
well?  Any thoughts on the media sent from the b-leg back to the a-leg when
it's not being expected (because there is no SDP)?


- Jeff



On Tue, Sep 24, 2013 at 11:03 PM, Muhammad Shahzad Shafi <
shahzad at voip-demos.com> wrote:

> **
>
> Well, you have to sacrifice 183 Early Media, since converting 183 to 180
> is far more easy and convenient then converting 180 to 183 (since then you
> have to involve a media server, which is not going to be so easy).
>
> Therefore, my advice would be to change all 183 from that carrier to 180
> response. You can use change_reply_status method,
>
>
> http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status
>
> Thank you.
>
>
>
> On 2013-09-25 03:05, Jeff Pyle wrote:
>
> No takers?  :)
>
> I wonder if it's possible to script this in a B2BUA scenario?  I'm not
> sure how one would do detection of 180 without SDP versus 180/183 with SDP
> in B2B-land.  Or, what to do from there once it knew.
>
>
>  - Jeff
>
>
>
> On Mon, Sep 23, 2013 at 10:43 AM, Jeff Pyle <jpyle at fidelityvoice.com>wrote:
>
>> Hi Laszlo,
>>
>> Unfortunately the effect for the caller would be the same - ringback
>> would stop.
>>
>> Here's the whole flow.  My terminating gateway is SIP to ISDN PRI.  Call
>> terminates through the gateway to a particular mobile switching office.  I
>> receive an ISDN PROGRESS message with inband audio.  This translates to the
>> 183 with SDP.  Then I receive an ALERTING message with no inband audio.
>>  This translates to the 180.  When the MSO sends the ALERTING, it has
>> stopped sending the inband audio from the previous PROGRESS message.
>>
>> I'm thinking I need to do something else in the onreply_route to connect
>> to the media server for a new 183.  Since I've executed t_relay to route
>> the INVITE to the gateway, it seems my options are limited.
>>
>>
>> - Jeff
>>
>>
>>
>> --
>> Jeff Pyle <jpyle at fidelityvoice.com>
>> Director, Voice Engineering
>> Fidelity Voice and Data
>> 216-245-4106
>> www.fidelityvoice.com
>>
>>
>>
>> On Mon, Sep 23, 2013 at 8:57 AM, Laszlo <laszlo at voipfreak.net> wrote:
>>
>>>  What if you simply drop the 180 in the onreply_route?
>>>
>>> -Laszlo
>>>
>>>
>>> 2013/9/23 Jeff Pyle <jpyle at fidelityvoice.com>
>>>
>>>>  Hello,
>>>>
>>>> I have one particular PSTN call flow that causes a 183 with SDP, then a
>>>> 180 without SDP prior to 200 OK.  Some of my customer endpoints don't
>>>> handle the 180 properly after a 183 and they cease to hear ringback.
>>>>
>>>> I'm thinking through how intercept the 180 and convert it to a 183 with
>>>> SDP.  I have a media server available to generate the 183 and the media.
>>>>  I'm struggling with how to relay the INVITE to the media server when the
>>>> 180 arrives in the middle of the call setup.
>>>>
>>>> Any recommendations are appreciated.
>>>>
>>>>
>>>>
>>>> Regards,
>>>> Jeff
>>>>
>>>>   _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>> --
>>>
>>> --
>>> Kind regards,
>>> Laszlo Bekesi
>>> http://voipfreak.net
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>      --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
>
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>
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