[Spce-user] sip fragmentation
Raúl Alexis Betancor Santana
rabs at dimension-virtual.com
Thu Apr 14 01:30:07 EDT 2016
You could only reduce the packet size using headers compresions, reuducing SDP offers or a combination of both but I'm sure they will not support that. VoIPStunt it's not a 'serious' carrier from you could spect support from their side on this.
Also it's strange that you packet get oversized if you have only 2 RR
About the no RTP-flow on internal calls ... check using sngrep the full-flow and see whats getting wrong when negotiated.
> 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: Jueves, 14 de Abril 2016 4:26:53
> Asunto: Re: [Spce-user] sip fragmentation
> I applied that 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 184.108.40.206.1 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: [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
>>> Via: SIP/2.0/UDP x.x.x.115;branch=z9hG4bK037e.d0ff913568a8906a194cc8f2492a77c7.0
>>> Via: SIP/2.0/UDP
>>> 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, 220.127.116.11 is VoIPStunt's
>>>>> ip address, the other is my spce's.
>>>>> Spce-user mailing list
>>>>> Spce-user at lists.sipwise.com
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