[Spce-user] Rings once, then disconnects!

[Andreas Granig]

Hi,

I guess it wouldn't be too hard to add filtering video, there is
http://www.kamailio.org/docs/modules/4.3.x/modules/sdpops.html#sdpops.f.sdp_remove_media
for this purpose.

You can try it in BRANCH_ROUTE_SBC of
/etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2 right before
the log line "Request leaving server".

Andreas

On 02/05/2016 07:40 AM, William Hilsum wrote:
> Hi Alex,
> 
> Apologies - I think I'm unintentionally being confusing!
> 
> This is not a problem as such with Sipwise - it is the sip client with the option "Allow video calls" that when turned on, it automatically causes my upstream carrier to reject the call (after a single ring, which is just weird...)
> 
> Anyway, I have another PBX here where on a per carrier basis, I have the option "Disable video calls". When this is ticked, it basically strips out all relevant headers such as what caused my calls on Sipwise to fail.
> 
> So my question really is, after finding out the issue here, is there anyway I can get Sipwise to strip out these headers to avoid similar issues? I have not tested, but, I am guessing if anyone has a VOIP videophone, they may have the same issue as the softphone wasn't actually trying to establish a video stream, it was just saying it was video compatible in the header.
> 
> Thanks,
> 
> William
> 
> -----Original Message-----
> From: Spce-user [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Alex Lutay
> Sent: 02 February 2016 08:24
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Rings once, then disconnects!
> 
> Dear William,
> 
> Check incoming invite from softphone to Sipwise, I am strongly believe you will find the same part there.
> 
> Sipwise doesn't change SDP part a lot, mainly it is replacing IP (I believe you have replaced '<PBX-IP>' in SDP part manually, otherwise it is an error) and limit list of codecs if necessary (turned off by default).
> 
> You can also try to make a call from your softphone through your Carrier directly and compare invite sent by softphone and Sipwise.
> 
> On the first look I do not see anything which can cause 500 on carrier side here. Was the correct IPv4/IPv6 in <PBX-IP> ?
> 
> On 02/02/2016 01:03 AM, William Hilsum wrote:
>> These additional lines in the "invite (SDP) packet" from Sipwise to the carrier platform break the connection:
>>
>> m=video 32562 RTP/AVP 34
>> c=IN IP4 <PBX-IP>
>> a=rtpmap:34 H263/90000
>> a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
>> a=sendrecv
>> a=rtcp:32563
>> a=direction:both
> 
> --
> Alexander Lutay
> Head of Quality Assurance
> Sipwise GmbH, Campus 21/Europaring F15
> AT-2345 Brunn am Gebirge
> 
> Office: +43(0)13012036
> Email: alutay at sipwise.com
> Website: http://www.sipwise.com
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> https://lists.sipwise.com/listinfo/spce-user
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> https://lists.sipwise.com/listinfo/spce-user
>