(not trying to hijack matthew's thread, but we answer both of the last 2 posts on this thread with this)
the constant being the spectralink look at frame 6 in the merged file matthew sent in. what is missing is that there is no ACK back. noone knows there is any media "really established" because the UA (spectralink) wont ack it.After seeing session progress, there are no ack's FROM the UA at 10.81.3.5) back to sipxbridge or the proxy at 172.30.216.62. OK, I keep saying OK to myself long enough and you dont say anything back I'm guessing you ain't there. It's a bad idea to leave this nailed up if you are there because if you were you's ACK me back. BYE (frame 18). it's a bug that needs to be fixed by polycom in the spectralink (IMO). Getting them to acknowledge and do anything about it is another thing altogether. On Mon, Aug 1, 2011 at 2:56 PM, McIlvin, Don <[email protected]>wrote: > Matthew,**** > > ** ** > > Anything new!**** > > ** ** > > I am getting the external call drop at 15 seconds too, multiple scenarios. > For us it is after dialing inbound.**** > > ** ** > > This is on a test environment.**** > > ** ** > > We tried a desktop Bria3 (just updated) calling extension to extension with > a remotely registered 3G iPhone using the Bria iPhone client. Call dropped > at 15 seconds.**** > > The goal of the test was to take Polycom and our carrier out of the > equation. So we still get the exactly 15 sec drop with no Polycom phone or > ITSP carrier in the mix. **** > > ** ** > > In contrast the same two end points, with the iPhone (using a Bria SIP > client) registered over the local WiFi in our building. Worked like a charm, > calls connect, two way audio, did not drop on call exceeding 2 minutes.*** > * > > ** ** > > We have a new SBC being tested (Audiocodes MSBG 6.2), but your > configuration seems to be using just SipX bridge.**** > > ** ** > > We also have a one way and sometimes zero way audio problem we are chasing > as well. On this, when we have one way audio it is the UDP RTP stream coming > in from the external device that does not get through. We have identified > that the BGP routers are not getting the udp/rtp stream sent to them, and > the FW sees no packets coming in (nor dropped).**** > > ** ** > > Don**** > > ** ** > > *From:* [email protected] > [mailto:[email protected]] *On Behalf Of *Matthew > Kitchin (public/usenet) > *Sent:* Monday, August 01, 2011 11:37 AM > *To:* sipx-users > *Subject:* [sipx-users] Fwd: Re: Polycom Spectralink 8020 - drops at 15 > seconds**** > > ** ** > > Polite bump request. > Does anyone see anything obvious in the sip trace? > > I'm 99.9% positive this has not been occurring all the time. If it has, my > testing is absolutely awful. > Outbound calls drop at 15 seconds. > This is the latest version of Sipx 4.4 with all updates. The server in this > example is virtual, but it is happening on all my physical boxes as well. > 8020 = 10.81.3.253 > Sipx = 10.81.3.5 > Verizon VoIP = 172.30.216.x > There is no IP address NAT, but there is an inbound 5060=>5080 NAT. > > > -------- Original Message -------- **** > > *Subject: * > > Re: [sipx-users] Polycom Spectralink 8020 - drops at 15 seconds**** > > *Date: * > > Thu, 28 Jul 2011 14:46:17 -0500**** > > *From: * > > Matthew Kitchin (public/usenet) > <[email protected]><[email protected]> > **** > > *To: * > > [email protected]**** > > > > 1 way audio was a false alarm. The phone was fine after a rebbot. The > dropped call at 15 second issue still exists on external calls. I am using > sipxbride. The sip trace is attached. The test call is the only call in the > logfiles. Any ideas? > > On 7/28/2011 2:41 PM, Tony Graziano wrote: **** > > "verizon is not seeing an ack on the call and thinks rtp is not actually > established." > > is now "sipxecs" is not verizon is not seeing an ack on the call and thinks > rtp is not actually established.**** > > On Thu, Jul 28, 2011 at 3:33 PM, Matthew Kitchin (public/usenet) < > [email protected]> wrote:**** > > I just realized they are getting 1 way audio even on internal calls too. > Something has to be wrong on our end. I will keep investigating. **** > > > On 7/28/2011 2:21 PM, Tony Graziano wrote: **** > > verizon is not seeing an ack on the call and thinks rtp is not actually > established. **** > > ** ** > > so then either the proxy is not sending the ack or the phone is not.**** > > ** ** > > got a siptrace?**** > > On Thu, Jul 28, 2011 at 3:15 PM, Matthew Kitchin (public/usenet) < > [email protected]> wrote:**** > > Can anyone help me interpret the attached wireshark? > I'm 99.9% positive this has not been occurring all the time. If it has, my > testing is absolutely awful. > Outbound calls drop at 15 seconds. > This is the latest version of Sipx 4.4 with all updates. The server in this > example is virtual, but it is happening on all my physical boxes as well. > 8020 = 10.81.3.253 > Sipx = 10.81.3.5 > Verizon VoIP = 172.30.216.x > There is no IP address NAT, but there is an inbound 5060=>5080 NAT. > I'm not an expert at reading these, but I believe the BYE is coming from > Verizon and I do not know why. I think I am going to have to open a ticket > with them, but I wanted to check here first and see if I am missing > something. > > Thanks, > Matthew > > _______________________________________________ > 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: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > 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 voip fax services!**** > > ** ** > > > > **** > > _______________________________________________**** > > sipx-users mailing list**** > > [email protected]**** > > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > > _______________________________________________ > 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: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > 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 voip fax services!**** > > > > > > **** > > _______________________________________________**** > > sipx-users mailing list**** > > [email protected]**** > > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > "The information in this electronic mail message is the sender's > confidential business and may be legally privileged. It is intended solely > for the addressee(s). Access to this internet electronic mail message by > anyone else is unauthorized. If you are not the intended recipient, any > disclosure, copying, distribution or any action taken or omitted to be taken > in reliance on it is prohibited and may be unlawful." > > "The sender believes that this E-mail and any attachments were free of > any virus, worm, Trojan horse, and/or malicious code when sent. This message > and its attachments could have been infected during transmission. By reading > the message and opening any attachments, the recipient accepts full > responsibility for taking protective and remedial action about viruses and > other defects. The sender's employer is not liable for any loss or damage > arising in any way from this message or its attachments." > > "In connection with representing sellers and/or buyers in real estate > transactions, Coldwell Banker Residential Brokerage real estate sales > associates have absolutely no authority to create binding contractual > obligations on behalf of a seller or on behalf of a buyer via any written or > verbal communications including, but not limited to email communications." > [v1.0.07.109] > > _______________________________________________ > 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: 434.326.5325 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 voip fax services!
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