[Freeswitch-users] SIP 100 response and media

[Steven Ayre]

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> 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> 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
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>>
>>
>> _______________________________________________
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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, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
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>
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