[Spce-user] Rings once, then disconnects!

William Hilsum William at ezpcltd.com
Fri Feb 5 01:40:48 EST 2016


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



More information about the Spce-user mailing list