[Freeswitch-users] SIP 100 response and media

Boris Kovalenko boris at tagnet.ru
Thu Jun 30 16:27:07 MSD 2011


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 
> <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
>>
>>
>>
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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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-- 
Regards,
Boris


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