[Freeswitch-users] SIP 100 response and media
Boris Kovalenko
boris at tagnet.ru
Thu Jun 30 18:54:21 MSD 2011
Hello!
Thank You, Steve!
> Yes, either that or 200 (you can skip ringing to go straight to being
> answered).
>
> Generally with 183/180 you'll get ringback but not hear voice in early
> media, you wouldn't normally hear voice until the 200 OK answers the call.
>
> Media can't start up without SDP in the body of the 180/183/200. SDP
> contains the IP/port used for media and the selected codec. Media
> can't work without those details. Sometimes you'll see 180/183 without
> SDP there'll still be no early media containing ringback, in that case
> the client'll usually generate its own ringback but you'll always see
> SDP in the 200 OK.
>
> -Steve
>
>
>
>
> On 30 June 2011 13:00, Boris Kovalenko <boris at tagnet.ru
> <mailto:boris at tagnet.ru>> wrote:
>
> Hello!
>
> But even reading this RFC, I found that 100 Trying should be
> followed by one of 18x response to get media channels connected,
> isn't?
>
>> 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
>> <mailto: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
>>> <mailto: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
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
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>>>
>>>
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>>
>>
>> --
>> С уважением,
>> Борис Коваленко
>> ЗАО "Тагнет"
>> тел. +7 (3435) 230001
>> факс +7 (3435) 230005
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
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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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>
>
>
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>
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--
Regards,
Boris
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