[Freeswitch-users] SIP 100 response and media

[Kerem Erciyes]

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
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
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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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>



-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com