[SIPForum-discussion] SIP Forum solution on FOIP T38 issues

[Tristan Bulandus]

Team, any news on SIP Forums announcment on solutions and recommendation on inter-op problems for T38?



________________________________
From: Tristan Bulandus <tristanbulandus at yahoo.com>
To: Saurabh Shah <saurabh.shah at matrixcomsec.com>; Heather Mayhan <heather.mayhan at ericsson.com> 
Cc: "discussion at sipforum.org" <discussion at sipforum.org>; AOEKING <aoeking at gmail.com> 
Sent: Wednesday, December 25, 2013 9:23 PM
Subject: Re: [SIPForum-discussion] CALL for HELP: Dead Air problem



Thanks team, issue has now been resolved, IP connection is okay, Firewall detected in between was blocking RTP ports. Now its okay. Thanks.





On Tuesday, December 24, 2013 12:33 PM, Saurabh Shah <saurabh.shah at matrixcomsec.com> wrote:

Please provide network setup. 

If router are part of your network, check port forwarding and NAT types in each router.

Regards,
Saurabh Shah

________________________________

From: "Heather Mayhan" <heather.mayhan at ericsson.com>
To: "AOEKING" <aoeking at gmail.com>, "Tristan Bulandus" <tristanbulandus at yahoo.com>
Cc: discussion at sipforum.org
Sent: Wednesday, December 18, 2013 8:55:49 PM
Subject: Re: [SIPForum-discussion] CALL for HELP: Dead Air problem


What protocol are you using to handle the media gateway component/media setup and tear-down?  
 
If it is Megaco, there should be messaging when the call releases (MEGACO SUBTRACT message) that should show packet statistics (packets sent, received, packet loss, jitter, delay, etc).  Normally when I hear of dead-air issues, that’s the first thing I look for.  Our CDR’s actually get a copy of this information, so I don’t always have to capture the Megaco to see it, but it depends upon what kind of set up you have.
 
 
From:discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of AOEKING
Sent: Monday, December 16, 2013 08:15
To: Tristan Bulandus
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] CALL for HELP: Dead Air problem

Hi,

Take a pcap trace and check whether the SDP part of the sip INVITE contains the remote global IPs.
I believe the IP phone uses the local IP's instead of using global IPs

Thanks
Nuwan 


On Sun, Dec 8, 2013 at 7:30 AM, Tristan Bulandus <tristanbulandus at yahoo.com> wrote:
Team, hoping for your help please. Thanks
 
On Saturday, December 7, 2013 6:40 PM, Tristan Bulandus <tristanbulandus at yahoo.com> wrote:
Hi Team,
 
May I once again call upon the SIP Experts from this international forum.
 
I am the pioneer of advanced SIP technologies in our telco, and I am currently facing a very difficult challenge in a new project involving SIP Clients, SIP Proxies, SIP Platfroms, etc. This is the first time I have experienced this problem and nobody locally and internationally that I work with is able to solve the problem.
 
The scenario is that I have IP Phones located at the site and RTP is two way but when I have IP Phones located on a remote site with IP connectivity there is no Media on both parties, for both inbound and outbound calls.
 
Appreciate if you can help me immediately.
 
Thanks.
 

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