[VoiceOps] Allworx Followme - no audio issue
ujjval at simplesignal.com
Fri Jul 22 19:35:25 EDT 2011
Thx for all for your responses guys.
After looking through the SIP signaling nothing seem to be wrong, all
reinvites are acknowledged with 200OKs and ACKs and no RTP ports are
changed mid-call by any device in this scenario..However, one odd thing
that came to light is that after the call is answered and connected the
RTP packets coming from Allworx PBX have a very low TTL of only 9. Until
call is answered, the RTP packet TTL coming from Allworx is at 126. That
is the only straw I have been able to grasp in this whole investigation,
as my carrier is telling me they see 2 way Audio as well on their
connection to us.
Anyone else seen similar issues due to low TTL values in RTP packets.
From: Justin Randall [mailto:lists at outofservice.org]
Sent: Tuesday, July 19, 2011 11:36 AM
To: Alex Balashov; Ujjval Karihaloo; voiceops at voiceops.org
Subject: Re: [VoiceOps] Allworx Followme - no audio issue
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,
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
> 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
>> the call.
>> PSTN cell phone presses 1 to answer the call following which there is
>> 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
>> 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?
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
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