[Freeswitch-users] SIP 100 response and media

[Steven Ayre]

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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