[Freeswitch-users] OneWay audio after hold->unhold

Tue Sep 8 13:24:03 MSD 2015

This is what I have got from my switch engineer.

Avaya PBX Version Control
§  Media encryption over IP:Y
§  Session Manager Element Manager 6.3
§  System Manager Release 6.3.5
§  Communication System Management 6.3.10
§  SIP Phones: Version 2.6.5.1 or higher (2.6.9-2.6.13)
§  One-X SoftPhone: Version 6.1.9.04-SP9 & 6.2.6.03-FP6

In System-Parameters Customer Options
§  Media encryption over IP:Y
§  Initial Invite with SDP for secure calls: Y

In IP-Network-region,
§  Codec set matches group for encryption

§  Intra-region IP-IP Direct Audio: yes

§  Inter-region IP-IP Direct Audio: yes

§  IP Audio Hairpinning: y

In IP-Codec Set, Media Encryption with
§  1-srtp-aescm128-hmac80” on first line for SRTP encryption
§  Aes on 2nd for conferencing

In CM, signaling-group of type: SIP
§  Transport Method: TLS
§  Enforce SIPS URI for SRTP?: N.
§  Peer Detection Enabled?: Y
§  Peer Server: Set to SM
§  Near-end node name: PROCR.
§  Near-end Listen Port: 506x
§  Far-end node name: SM.
§  Far-end Listen Port: 506x
§  Direct IP-IP Audio Connections?: Y
§  IP Audio Hairpinning?: Y
§  Initial IP-IP Direct Media?: Y

In CM, Trunk Type:,
§  Trunk Type: SIP
§  Service Type: Tie
§  Direction: 2 way
Protocol Variations:
o   Direction: 2 way
o   Send Transferring Party Information? Y
o   Network Call Redirection? Y
o   Convert 180 to 183 for Early Media? N
o   Identity for Calling Party Display: P-Asserted-Identity
o   Interworking of ISDN Clearing with In-Band Tones: keep-channel-active

Session Manager:
                Adaptations Required
§  Fromto: True
§  Iosrcd: domain
§  Iodstd: domain

Prashanth

From: Bote Man [mailto:bote_radio at botecomm.com]
Sent: 07 September 2015 23:03
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold

Actually, yes I need all of the settings on the Avaya side.

I am not using encryption and I never could get the Avaya CS1000M to talk to the FreeSWITCH installation. I think the closest I could get was the signaling would allow the call to be answered, but I got no audio to pass either direction.

I think the Avaya puts information in unexpected elements of the SIP message and I had no more time to troubleshoot it because it was not my primary task.

There must be some small setting that I am missing and I would love to find the missing piece of the puzzle.

Thanks!

Bote

From: Prashanth Devarajappa
Sent: Monday, 07 September, 2015 09:47
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold

Avaya PBX Version Control
•  Media encryption over IP:Y
•  Session Manager Element Manager 6.3
•  System Manager Release 6.3.5
•  Communication System Management 6.3.10

In IP-Network-region :
•  Codec set matches group for encryption

•  Intra-region IP-IP Direct Audio: yes

•  Inter-region IP-IP Direct Audio: yes

In CM, signaling-group of type: SIP
•  Transport Method: TLS

Let me know if you need further Avaya config details.

FreeSwitch :
Its standard FS with custom (code/config) changes to enable

•         Support for RFC 5939 : Capability Negotiation

•         Support for UNENCRYPTED_SRTCP

•         Support IP Shuffling with in ACK…… to be done ☺

Prashanth

From: Bote Man [mailto:bote_radio at botecomm.com]
Sent: 05 September 2015 04:50
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold

We would be VERY appreciative if you could provide the configurations on the Avaya and FreeSWITCH that allow them to talk to each other. In my experience Avaya does not play well with others, and I have tried and given up.
Thank you in advance!
Bote

On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa <Prashanth.Devarajappa at enghouse.com<mailto:Prashanth.Devarajappa at enghouse.com>> wrote:
Hello,

I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I try hold -> unhold operation from FS for an outbound call.

When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints). FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection info(c=).

On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio.

Has anyone come across this issue ?

Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ?

Regards
Prashanth

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