[Spce-user] sip fragmentation

[Jonathan Yue]

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>
To: 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>
>>Para: "Raúl Alexis Betancor Santana" <rabs at dimension-virtual.com>, 
>>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 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>
>>To: 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>
>>>>Para: 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
>>>>
>>>>_______________________________________________
>>>>Spce-user mailing list
>>>>Spce-user at lists.sipwise.com
>>>>https://lists.sipwise.com/listinfo/spce-user
>>
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