[Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header

[Michal Zubáč]

Hi.

I tried all the things you advised me, but with no effect. When provider 
send us re-INVITE, FreeSwitch doesn't react to it at all.

Traces for test call are here:
   PCAP: http://et5.comgate.cz/zubacm/voice/dt_10m_4_sip.zip
   TEXT SUMMARY: http://pastebin.com/BgPUfmuc

I looked into FS source code and I didn't see anything what looks like 
response to SIP re-INVITE message. It looks like FS can't process any of 
incoming RFC 4028/3311 mechanisms.


I solved this issue by instructing FS to send re-INVITEs from our side 
before provider does it, effectively avoiding processing provider's 
re-INVITEs.

Michal Zubac
ComGate Interactive s.r.o.


On 23.1.2012 20:57, Kristian Kielhofner wrote:
> Michal,
>
>    I can barely read these parsed out SIP traces (yuck) but from the
> skimming I did it looks like you'd benefit from a few changes:
>
> - Explicitly disable session timers in your Sofia profile
> - Set  ignore_display_updates=true (so FreeSWITCH doesn't send an
> UPDATE - won't help in this case but a good thing to do)
> - Request your provider uses re-INVITEs instead of UPDATE
>
> On Mon, Jan 23, 2012 at 11:24 AM, Michal Zubáč<michal.zubac at comgate.cz>  wrote:
>> I don't have "enable-timer" or "session-timeout" variables set in Sofia
>> profile, so maybe defaults? I think these are what you are referring to.
>>
>> SIP trace is at
>>    http://pastebin.com/DTQU3nMr
>>
>> Yes, provider offered us re-INVITE method as an alternative for UPDATE.
>> Would this one work? What do I have to set up?
>>
>> Michal Zubac
>> ComGate Interactive s.r.o.
>>
>>
>> On 23.1.2012 15:51, Kristian Kielhofner wrote:
>>> What are your Sofia profile session timer values set to?
>>>
>>> Can you post a complete SIP trace?
>>>
>>> Can your provider send a re-INVITE instead of UPDATE?
>>>
>>> On Mon, Jan 23, 2012 at 8:40 AM, Michal Zubáč<michal.zubac at comgate.cz>    wrote:
>>>> Hello.
>>>>
>>>> I'd like to remove UPDATE value from SIP Accept header when creating SIP
>>>> calls. We're sending it in INVITE message and our provider uses that for
>>>> in-call keep-alive checks every 10 minutes.
>>>> FreeSwitch doesn't respond to that, so our provider disconnects RTP and
>>>> call is dropped. According to RFC3311 we can indicate that we don't
>>>> support this by not sending UPDATE in Accept header. Is this gonna help?
>>>>
>>>> Is there any way to drop that from SIP headers from dialplan? Or do I
>>>> have to change source code?
>>>> Or better, is there any other (cleaner) way to resolve this?
>>>>
>>>> Regards
>>>>
>>>> Michal Zubac
>>>> ComGate Interactive s.r.o.
>>>>
>>>> _________________________________________________________________________
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>>>>
>>>> 
>>>> 
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>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>
>