[Spce-user] sip fragmentation

[Andreas Granig]

Hi,

Fragmentation issues typically occur when a firewall is broken and drops
the fragmented parts, or if the operating system of the remote device is
broken and can't re-assemble them (although the former issue is much
more frequent).

One way to solve it is to use TCP instead of UDP.

Andreas

On 04/14/2016 05:26 AM, Jonathan Yue wrote:
> I applied that workaround:
> https://www.sipwise.org/doc/2.8/spce/ar01s05.html#_audiocodes_devices_workaround,
> from the packet captures, I can see 127.0.0.1 in RR header is changed to
> a private ip address. but issue persists.
>  
> So I went back to the thought of reducing sdp size. on softphone
> MicroSIP, I disabled ICE, enabled only 1 codec. bingo! the sdp is now
> small enough to be contained in 1 packet. then the outbound call via
> VoIPStunt succeeded! this test proved my thought about packet
> fragmentation, but I still wonder what is the official solution for such
> situation? to disable ICE and keep only 1 codec isn't nice way to
> configure a sip client.
>  
> Now the new problem is that there's no rtp traffics between in internal
> call.
>  
> ------ Original Message ------
> From: "Raúl Alexis Betancor Santana" <rabs at dimension-virtual.com
> <mailto:rabs at dimension-virtual.com>>
> To: spce-user at lists.sipwise.com <mailto:spce-user at lists.sipwise.com>
> Sent: 2016-04-13 4:11:29 PM
> Subject: Re: [Spce-user] sip fragmentation
>  
>> Trust me ... it's not a fragmentation problem ... just follow the
>> indications on the documentation about the "Audiocodes devices
>> workaround", the real problem it's the 1.27.0.0.1 IP on the RR headers
>> ... VoIPStunt and a lot of other SIP-Carriers, doesn't support private
>> IP's on the SIP headers.
>>
>>
>> ------------------------------------------------------------------------
>>
>>     *De: *"Jonathan Yue" <jonathan.yue at turboitsolutions.com
>>     <mailto:jonathan.yue at turboitsolutions.com>>
>>     *Para: *"Raúl Alexis Betancor Santana" <rabs at dimension-virtual.com
>>     <mailto:rabs at dimension-virtual.com>>, spce-user at lists.sipwise.com
>>     <mailto:spce-user at lists.sipwise.com>
>>     *Enviados: *Miércoles, 13 de Abril 2016 23:09:45
>>     *Asunto: *Re[2]: [Spce-user] sip fragmentation
>>
>>     Thanks for your input Raúl.
>>      
>>     in the packet captures, I notice following items:
>>     INVITE sip:1xxxxxxxxxx at sip.voipstunt.com
>>     <mailto:1xxxxxxxxxx at sip.voipstunt.com> SIP/2.0
>>     Record-Route:
>>     <sip:x.x.x.115;r2=on;lr=on;ftag=3F30077D-570BCE0A000B61B3-54909700;ngcplb=yes>
>>     Record-Route:
>>     <sip:127.0.0.1;r2=on;lr=on;ftag=3F30077D-570BCE0A000B61B3-54909700;ngcplb=yes>
>>     Via: SIP/2.0/UDP
>>     x.x.x.115;branch=z9hG4bK037e.d0ff913568a8906a194cc8f2492a77c7.0
>>     Via: SIP/2.0/UDP
>>     127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsdiM4aEk;rport=5080
>>      
>>     I wonder if the 2nd record-route message and 2nd Via message are
>>     necessary. If they are omitted, the sip packet won't be fragmented.
>>      
>>     ------ Original Message ------
>>     From: "Raúl Alexis Betancor Santana" <rabs at dimension-virtual.com
>>     <mailto:rabs at dimension-virtual.com>>
>>     To: spce-user at lists.sipwise.com <mailto:spce-user at lists.sipwise.com>
>>     Sent: 2016-04-13 1:02:40 AM
>>     Subject: Re: [Spce-user] sip fragmentation
>>      
>>
>>         You could activate header compresion, but you will have to
>>         pray to  VoIPStunt to support that.
>>
>>         Better check why your headers get so big
>>
>>         ------------------------------------------------------------------------
>>
>>             *De: *"Jonathan Yue" <jonathan.yue at turboitsolutions.com
>>             <mailto:jonathan.yue at turboitsolutions.com>>
>>             *Para: *spce-user at lists.sipwise.com
>>             <mailto:spce-user at lists.sipwise.com>
>>             *Enviados: *Martes, 12 de Abril 2016 23:48:43
>>             *Asunto: *[Spce-user] sip fragmentation
>>
>>             Hi, gang,
>>              
>>             Is there a way to shrink the header of sip message sent
>>             from spce? I configured a sip trunk with VoIPStunt as a
>>             test. After receiving invite message, VoIPStunt sends 401,
>>             spce then sends out an sdp, supposedly containing
>>             authentication digest, however, the message gets
>>             fragmented. VoIPStung never responds again, apparently it
>>             doesn't assemble fragmented packets.
>>              
>>             I then change the trunk's transport protocol to tcp,
>>             VoIPStunt doesn't respond at all, I guess it doesn't
>>             support tcp.
>>              
>>             spce version is v4.2.1. in the attached screenshot,
>>             77.72.169.129 is VoIPStunt's ip address, the other is my
>>             spce's.
>>              
>>             thanks,
>>              
>>             Jon
>>
>>             _______________________________________________
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>>             <mailto:Spce-user at lists.sipwise.com>
>>             https://lists.sipwise.com/listinfo/spce-user
>>
>>
> 
> 
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