[Freeswitch-users] SIP 100 response and media

[Boris Kovalenko]

Hello!

     Would You please tell me what IOS version have You used?
> Hi Boris,
>
> I had some trouble with 28xx series before, but I was trying to use it
> as a PSTN-2-SIP gateway. Try to see if the firmware is up-to-date, as
> there were some SIP protocol related glithces in the code fixed in
> newer firmware.
>
> Kerem
>
>
> On Thu, Jun 30, 2011 at 3:27 PM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      Have reread your post. So 100 Trying may be (or may be not) followed by
>> 180/183. And in case of 180/183 early media will be processed (one way
>> only?). And full media will be only after the 200 OK from Cisco side. It
>> this right? In other words - Cisco *MUST* respond with 100 Trying and 200 OK
>> at least?
>>
>>
>> Btw, this is the latest SIP spec, better than the draft spec for 183 you
>> posted before.
>>
>> http://www.ietf.org/rfc/rfc3261.txt
>>
>> -Steve
>>
>>
>>
>> On 30 June 2011 09:55, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> Hello!
>>>
>>> Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my
>>> knowledge of SIP if right. And customer lies to me or forgot to configure
>>> something :)
>>>
>>> Is that all the signalling?
>>>
>>> 100 Trying only acknowledges the INVITE.
>>>
>>> That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
>>> would only be able to send early media if that was a 183/SDP or 180/SDP.
>>>
>>> The voice will only start when it sends 200/SDP which is when the call is
>>> answered.
>>>
>>> -Steve
>>>
>>>
>>> On 30 June 2011 06:00, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>>> Hello!
>>>>
>>>>      I have a customers with Cisco 2821 which want to use IVR on it. He
>>>> has configured Cisco and told me there is no voice when number is
>>>> called. The configuration is:
>>>>
>>>> Cisco 2821<-- FreeSwitch (full proxy mode) -->  PSTN
>>>>
>>>> I looked at tcpdump and the call flow is:
>>>> FS (INVITE) -->  Cisco 2821
>>>> Cisco 2821 (100 Trying) -->  FS
>>>> Cisco 2821 (RTP) -->  FS
>>>>
>>>> I think the call flow is wrong in that case and SIP100 can not go
>>>> throught Freeswitch
>>>> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>>>> should be SIP180 to get it work. But customers said this is OK and he
>>>> has no troubles with same configuration with other provider. What may I
>>>> do in this sutiation? Any specific Freeswitch configuration?
>>>>
>>>> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>>>>
>>>> --
>>>> Regards,
>>>> Boris
>>>>
>>>>
>>>>
>>>> _______________________________________________
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>>>
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>>>
>>> --
>>> С уважением,
>>>    Борис Коваленко
>>>    ЗАО "Тагнет"
>>>    тел. +7 (3435) 230001
>>>    факс +7 (3435) 230005
>>>
>>> _______________________________________________
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>>>
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>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
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>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
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>>
>>
>
>


-- 
Regards,
Boris