Hello, I've actually done interop work between AllWorx and Acme Packet in the past and found one-way audio issues when AllWorx in a few scenarios due to the way they behave in the face of re-INVITEs and having Acme configured to handle NAT correctly. If I recall correctly it changes its RTP port with every re-INVITE. With Acme configured for media latching - a key feature to assist with RTP through NAT (ignore SDP and "latch-on" to the first incoming RTP packet as the correct RTP destination). What ends up happening is that by the time the Acme generates the ACK to the 200 OK, before AllWorx has a chance to process the ACK and start sending using the new RTP port, Acme is still receiving media and "latches on" to the same IP/port being used before the re-INVITE, causing one-way audio where the AllWorx side can't hear inbound audio. The fix for this would be one of the following: - Configure AllWorx to not change its RTP port with every re-INVITE. (I don't believe this is possible) - Put the AllWorx system on its own Acme realm with restrictive latching disabled and do not place the AllWorx system behind NAT. Hope this helps, Justin Randall On 7/19/11, Alex Balashov <abalashov at evaristesys.com> wrote:
This is little more than a descriptive restatement of your original post. What is the SIP flow?
If there is two-way audio loss after the cell phone accepts the call, there must be a reinvite that is changing the SDP or some other event of significance associated with that. Most likely, the problem lies precisely in the nature of that endpoint pivot, e.g. the new, revised RTP endpoints do not have direct network and transport-layer reachability to each other, or are otherwise unable to fulfill the request.
On 07/19/2011 12:49 PM, Ujjval Karihaloo wrote:
Call from PSTN to our Broadsoft/ACME to Customer SIP Trunk/Allworx
Allworx forwards/followme call back out (to a cell phone) the same SIP trunk to us and we terminate to the PSTN (cell phone)
PSTN cell phone answers and Allworx plays announcement - press 1 to answer the call.
PSTN cell phone presses 1 to answer the call following which there is no audio in either direction.
In the packet capture between Allworx and my ACME (access SIDE) - I see RTP flowing, I also see RTP flowing between ACME and my Upstream carrier who terminated the call..and inbound carrier who delivered the inbound call to me.
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Alex Balashov Sent: Tuesday, July 19, 2011 10:43 AM To: voiceops at voiceops.org Subject: Re: [VoiceOps] Allworx Followme - no audio issue
On 07/19/2011 12:40 PM, Ujjval Karihaloo wrote:
Hi All:
We have a customer with Allworx PBX that is setup for followme on an extension that forwards to a call Phone. When the call gets forwarded, back out the SIP Turnk intous, the allworx has answer confirmation which the followme/cellphone hears fine, when the followme cell phone presses 1 to accept the call, all audio is lost both ways. Call stays up until disconnected. The Allworx sends a REINVITE after the "1" is pressed.too
Anyone seen this behavior with ACME and Allworx and any remedies.
Well, what exactly is the SIP flow?
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