Issue when receiving INVITE with both audio and image (T38)
Hello list, I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending both audio+image (T38) in the initial INVITE. In a simple bridge scenario, FS will just ignore the T38 part and setup an audio connection only with b-leg. My issue is later on if/when we receive a re-INVITE (ex: SIP timers refresh) with the same SDP (audio+T38), FS now considers it as FAX and sends a re-INVITE to b-leg with T38 only! In case the destination is voice only, the call hangs up. --- Asterisk apparently had the same issue, they solved it by setting up both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL they just re-INVITE b-leg with T38. See: https://reviewboard.asterisk.org/r/208/ Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843 Is this the right way to go? (Also is it feasible in FreeSwitch)? I can add a JIRA if necessary. Thanks, François.
We can investigate that, Meanwhile you could use regex in your dialplan for this situation and set "sip_ignore_reinvites=true" to mitigate. On Fri, Aug 16, 2013 at 8:56 AM, François <fdelawarde@wirelessmundi.com>wrote:
Hello list,
I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending both audio+image (T38) in the initial INVITE. In a simple bridge scenario, FS will just ignore the T38 part and setup an audio connection only with b-leg.
My issue is later on if/when we receive a re-INVITE (ex: SIP timers refresh) with the same SDP (audio+T38), FS now considers it as FAX and sends a re-INVITE to b-leg with T38 only! In case the destination is voice only, the call hangs up.
---
Asterisk apparently had the same issue, they solved it by setting up both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL they just re-INVITE b-leg with T38.
See: https://reviewboard.asterisk.org/r/208/ Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843
Is this the right way to go? (Also is it feasible in FreeSwitch)? I can add a JIRA if necessary.
Thanks, François.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thanks for the tip. Should I add a JIRA ticket? François. On Fri, 2013-08-16 at 10:00 -0500, Anthony Minessale wrote:
We can investigate that, Meanwhile you could use regex in your dialplan for this situation and set "sip_ignore_reinvites=true" to mitigate.
On Fri, Aug 16, 2013 at 8:56 AM, François <fdelawarde@wirelessmundi.com> wrote: Hello list,
I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending both audio+image (T38) in the initial INVITE. In a simple bridge scenario, FS will just ignore the T38 part and setup an audio connection only with b-leg.
My issue is later on if/when we receive a re-INVITE (ex: SIP timers refresh) with the same SDP (audio+T38), FS now considers it as FAX and sends a re-INVITE to b-leg with T38 only! In case the destination is voice only, the call hangs up.
---
Asterisk apparently had the same issue, they solved it by setting up both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL they just re-INVITE b-leg with T38.
See: https://reviewboard.asterisk.org/r/208/ Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843
Is this the right way to go? (Also is it feasible in FreeSwitch)? I can add a JIRA if necessary.
Thanks, François.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Sure On Aug 16, 2013 11:14 AM, "François" <fdelawarde@wirelessmundi.com> wrote:
Thanks for the tip. Should I add a JIRA ticket?
François.
On Fri, 2013-08-16 at 10:00 -0500, Anthony Minessale wrote:
We can investigate that, Meanwhile you could use regex in your dialplan for this situation and set "sip_ignore_reinvites=true" to mitigate.
On Fri, Aug 16, 2013 at 8:56 AM, François <fdelawarde@wirelessmundi.com> wrote: Hello list,
I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending both audio+image (T38) in the initial INVITE. In a simple bridge scenario, FS will just ignore the T38 part and setup an audio connection only with b-leg.
My issue is later on if/when we receive a re-INVITE (ex: SIP timers refresh) with the same SDP (audio+T38), FS now considers it as FAX and sends a re-INVITE to b-leg with T38 only! In case the destination is voice only, the call hangs up.
---
Asterisk apparently had the same issue, they solved it by setting up both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL they just re-INVITE b-leg with T38.
See: https://reviewboard.asterisk.org/r/208/ Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843
Is this the right way to go? (Also is it feasible in FreeSwitch)? I can add a JIRA if necessary.
Thanks, François.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
There you go: http://jira.freeswitch.org/browse/FS-5714 Thanks, François. On Fri, 2013-08-16 at 13:01 -0500, Anthony Minessale wrote:
Sure
On Aug 16, 2013 11:14 AM, "François" <fdelawarde@wirelessmundi.com> wrote:
Thanks for the tip. Should I add a JIRA ticket?
François.
On Fri, 2013-08-16 at 10:00 -0500, Anthony Minessale wrote: > We can investigate that, Meanwhile you could use regex in your > dialplan for this situation and set "sip_ignore_reinvites=true" to > mitigate. > > > > > On Fri, Aug 16, 2013 at 8:56 AM, François > <fdelawarde@wirelessmundi.com> wrote: > Hello list, > > I've seen several gateways (Patton, Audiocodes, Nortel CS2K), > sending > both audio+image (T38) in the initial INVITE. In a simple > bridge > scenario, FS will just ignore the T38 part and setup an audio > connection > only with b-leg. > > My issue is later on if/when we receive a re-INVITE (ex: SIP > timers > refresh) with the same SDP (audio+T38), FS now considers it as > FAX and > sends a re-INVITE to b-leg with T38 only! In case the > destination is > voice only, the call hangs up. > > --- > > Asterisk apparently had the same issue, they solved it by > setting up > both streams on a-leg (RTP + UDPTL). If they receive packets > on UDPTL > they just re-INVITE b-leg with T38. > > See: https://reviewboard.asterisk.org/r/208/ > Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843 > > Is this the right way to go? (Also is it feasible in > FreeSwitch)? I can > add a JIRA if necessary. > > Thanks, > François. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Anthony Minessale
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François