[Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable

[Birgit Arkesteijn]

Hi Anthony,

I'm fairly sure that it's not an IAX problem. IAX2 messages are properly 
acked, sound (between IAX2 and SIP) is ok, etc, etc.

The problem is (only) with DTMF, as far as I can tell.
It's more likely to be in core or sofia_sip?

Thanks, Birgit


On 11/03/08 13:20, Anthony Minessale wrote:
> Just to warn you we do not use mod_iax all that much.  It was used as a 
> tool to test the core of FreeSWITCH in the early stages of development 
> and we just kinda left it hanging around since it does at least work.  
> We had to add a bunch of code to the stock iaxclient lib to make it 
> threadsafe etc because the lib is designed to handle 1 or 2 calls and we 
> are using it in a b2bua type env where hundreds of concurrent calls may 
> be possible.  That said, since we more or less forked the code to avoid 
> generating support issues for the other soft-phones that use iaxclient, 
> you may want to check if there were any bugs in the real version of the 
> library that match your issue.
>  
> http://sourceforge.net/projects/iaxclient
> 
> 
> We've had little to no demand for IAX2 so we have not focused very much 
> on it. We do have strategies for creating a more robust and perhaps even 
> an "improved" implementation of the IAX2 protocol. Alas, it's pretty 
> much purely an academic exercise because there is little reason to 
> implement it.
> 
> 
> 
> 
> 
> 
>  
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> 
> 
> ----- Original Message ----
> From: Birgit Arkesteijn <birgit at westhawk.co.uk>
> To: Freeswitch-users at lists.freeswitch.org
> Sent: Monday, March 10, 2008 10:52:40 AM
> Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
> 
> Hi,
> 
> We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64).
> 
> I'm experiencing problems with DTMF:
> The IAX2 DTMF frames aren't always transferred into RTP packets.
> Sometimes none of the frames are translated, sometimes only the first
> one is omitted. All the IAX2 DTMF packets are ack-ed.
> 
> Our setup:
> PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol)
> -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>)
> -> SIP to SystemX (sofia/default/123 at 66.212.134.192)
> 
> 
> I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug 
> that could explain this, but
> failed, so I submitted this as a bug, see
> http://jira.freeswitch.org/browse/FSCORE-109
> 
> Has anyone else experienced the same or a similar problem?
> 
> Thanks, Birgit
> 


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-- Birgit Arkesteijn, birgit at westhawk.co.uk,
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