[VoiceOps] Hairpin Call from certain PBXs - No Audio issue
David.Hiers at adp.com
Fri May 10 12:48:06 EDT 2013
If you can get something like TBCT working, you can release the media back upstream somewhere along the line.
Other than that, I suppose that you could jack the TTL with iptables. Tough call, because the audio delay after that many IP hops will probably trash the call.
From: VoiceOps [mailto:voiceops-bounces at voiceops.org] On Behalf Of Ujjval Karihaloo
Sent: Thursday, May 09, 2013 21:30
To: voiceops at voiceops.org
Subject: [VoiceOps] Hairpin Call from certain PBXs - No Audio issue
Have you all seen issues with hairpin calls from PSTN--> ITSP (Broadsoft/ACME)--> SIP Trunk - which in turn forwards the call back out the SIP trunk to the PSTN and there is no Audio in either direction (Transfer via Auto-attendant OR Blind - Attended Xfer works as Media is anchored/reinitiated on PBX)
Having worked on a few PBXs (Allworx, Zultys) that cause this issue on ACME/Broadsoft Sip trunk setup - we see RTP in both direction on our traces, but due to the pass through RTP nature of this call flow, the TTL get way down to 5-6 in some cases, and that is what I believe is the issue, and our upstream vendors like Level3, Bandwidth etc suggest the same.
Other than implementing a RTP repeater of sorts, I don't see an alternative. Ideas appreciated
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