If the invite is FROM your sipx server, did you change the RTP ports? The default would be 30000-31000, but the diagram below shows 20000 range. So, how about explaining what is controlling the PSTN connectivity? Is it a POTS/PRI gateway, siptrunk, etc.?
On Tue, Oct 18, 2011 at 7:47 AM, Philippe Bechamp < [email protected]> wrote: > Hi Tony, > > Thanks for the reply ! > > I didn't get a sip-trace but I will a bit later when I get in the office. > > In the mean time, I can offer Wireshark's VOIP Flow diagram. > > Thanks ! > > Philippe. > > |Time | 10.70.225.101 | > | | | 10.70.103.100 | > |15.526 | INVITE SDP (g722 g711U g711A g729 > telephone-ev...RTPType-101) |SIP From: "Philippe Bechamp" < > sip:[email protected] To:< > sip:[email protected];user=phone > | |(5060) ------------------> (5060) | > |15.527 | 100 Trying| |SIP Status > | |(5060) <------------------ (5060) | > |15.528 | 407 Proxy Authentication Required |SIP Status > | |(5060) <------------------ (5060) | > |15.554 | ACK | |SIP Request > | |(5060) ------------------> (5060) | > |15.557 | INVITE SDP (g722 g711U g711A g729 > telephone-ev...RTPType-101) |SIP From: "Philippe Bechamp" < > sip:[email protected] To:< > sip:[email protected];user=phone > | |(5060) ------------------> (5060) | > |15.558 | 100 Trying| |SIP Status > | |(5060) <------------------ (5060) | > |24.601 | 183 Session progress SDP (g711U > telephone-even...PType-101) |SIP Status > | |(5060) <------------------ (5060) | > |24.712 | RTP (g711U) |RTP Num packets:17 > Duration:0.320s SSRC:0x12FE0FDC > | |(2230) ------------------> (20006) | > |24.712 | RTP (g711U) |RTP Num packets:2374 > Duration:47.460s SSRC:0x12FE0FDC > | |(2230) <------------------ (20002) | > |24.720 | RTP (g711U) |RTP Num packets:32 > Duration:0.299s SSRC:0x28A943DA > | |(2230) <------------------ (20002) | > |25.041 | RTP (g711U) |RTP Num packets:4722 > Duration:47.198s SSRC:0x489F3B37 > | |(2230) <------------------ (20002) | > |25.051 | 200 OK SDP (g711U telephone-eventRTPType-101) > |SIP Status > | |(5060) <------------------ (5060) | > |25.053 | RTP (g711U) |RTP Num packets:2357 > Duration:47.120s SSRC:0x12FE0FDC > | |(2230) ------------------> (20006) | > |25.070 | ACK | |SIP Request > | |(5060) ------------------> (5060) | > |72.229 | BYE | |SIP Request > | |(5060) ------------------> (5060) | > |72.259 | 200 OK | |SIP Status > | |(5060) <------------------ (5060) | > > > > On 2011-10-18, at 6:56 AM, Tony Graziano wrote: > > what is your pstn connectivity? did you grab a siptrace? if so, I. an look > at that if you like. > On Oct 18, 2011 2:27 AM, "Philippe Bechamp" < > [email protected]> wrote: > >> Hi there, >> >> Wonder if anyone can help me troubleshoot a one way audio problem. >> >> First of all, everything works fine with Bria - I can call anywhere in any >> direction. >> >> The issue happens with Polycom IP450 phones. >> >> The setup had been working fine for weeks and weeks. Then one day, sound >> seems to progressively degrade when making outgoing calls. After a while, >> boom, the receiving party of calls originating from the IP450 hears us but >> we can't hear them. The problem lasted about 36 hours the went away on it's >> own or I am not sure what i did to make it go away. >> >> Now, two weeks later, this has happened again. >> >> I have updated the IP450 to the recommended version on the wiki. >> >> I have traced and traced. It does not seem to be a firewall/nat/helper >> issue as a) Bria works fine on the same subnet b) doing port copies on my >> switch, I can see the packets being sent to the hardphone. >> >> From my limited but growing understanding of SIP calls, the server -> >> phone direction seems doubled (though it might just look ok to me…). I can't >> hear a thing and the diagnostics / media statistics on the phone increment >> in the outgoing direction but not in the incoming direction. >> >> Anyone have any idea that could help me pin this down ? Might it just be a >> phone firmware issue ? >> >> I have supporting material if that can help. >> >> Thanks, >> >> Philippe. >> >> >> >> |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||**** >> >> Philippe Bechamp | Manager, IT / Security**** >> >> Neuralitic Systems Inc. >> >> >> M. +1.514.812.9609**** >> >> E. [email protected]**** >> >> This message and attachments are for the use of the addressee only and may >> be confidential. If you are not the intended recipient, please notify the >> sender immediately by e-mail and delete this message and attachments. Thank >> you.**** >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > <ATT00001.c> > > > |||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||**** > > Philippe Bechamp | Manager, IT / Security**** > > Neuralitic Systems Inc. > > > M. +1.514.812.9609**** > > E. [email protected]**** > > This message and attachments are for the use of the addressee only and may > be confidential. If you are not the intended recipient, please notify the > sender immediately by e-mail and delete this message and attachments. Thank > you.**** > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
