[Freeswitch-users] OneWay audio after hold->unhold
Prashanth Devarajappa
Prashanth.Devarajappa at enghouse.com
Tue Sep 8 16:21:21 MSD 2015
Hi Kaleem,
Below are my response
>> is there any reason you are going for SRTP?
Customer work in Financial sector. Hence its pre requisite.
>> I always advise customers to first setup normal link and fully test then move to SRTP.
You are right. We don’t see any such issue with RTP media. Things get complicated with Encryption/secure media but that’s mandatory requirement.
>> when you are saying freeswitch not responding with key, are you seeing the message in FS Debug being generated?
Correct. I see new key in debug logs.
>> then next step would be to see this in traceSM logs and see if SM100 is getting these and how is SM responding to these and if sending to CM or not.
I was able to identify this issue using TraceSM. As there is no defined response for ACK, FS not sending anything back to SM. Hence I wanted to know if I can trigger UPDATE or 200 OK with my initial email to the group.
Prashanth
From: kaleem rehman [mailto:k4kaleem at gmail.com]
Sent: 08 September 2015 12:52
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold
Hi Prashanth,
is there any reason you are going for SRTP?
I always advise customers to first setup normal link and fully test then move to SRTP.
when you are saying freeswitch not responding with key, are you seeing the message in FS Debug being generated?
if not then you will need to raise a JIRA.
if you can, then next step would be to see this in traceSM logs and see if SM100 is getting these and how is SM responding to these and if sending to CM or not.
thanks, kaleem
On Tue, Sep 8, 2015 at 11:08 AM, <freeswitch-users-request at lists.freeswitch.org<mailto:freeswitch-users-request at lists.freeswitch.org>> wrote:
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Today's Topics:
1. Re: OneWay audio after hold->unhold (Prashanth Devarajappa)
---------- Forwarded message ----------
From: Prashanth Devarajappa <Prashanth.Devarajappa at enghouse.com<mailto:Prashanth.Devarajappa at enghouse.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc:
Date: Tue, 8 Sep 2015 10:07:27 +0000
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold
Thanks for your message.
>> Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch
This is how we have setup our environment and got this far.
Prashanth
From: kaleem rehman [mailto:k4kaleem at gmail.com<mailto:k4kaleem at gmail.com>]
Sent: 08 September 2015 10:46
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 82
Hi Prashanth / Bob,
Avaya CM supports sip but this is mainly used for Aura VM, Experience portal and SM and more of Avaya only devices, direct conenctions to Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP Limitations and also officially this is not supported
Best Route to do this is via session Manager as this is the supported method.
First create a trunk link between Session manager and Freeswitch, then register extension directly to SM and make calls between from this extension and Freeswitch, once everything is working then create a trunk between CM & SM, or use existing and setup digit routing on SM and CM.
if you have any issues setting up SM and its entities then let me know. this way will be best and far most easy one.
thanks, kaleem
On Tue, Sep 8, 2015 at 9:54 AM, <freeswitch-users-request at lists.freeswitch.org<mailto:freeswitch-users-request at lists.freeswitch.org>> wrote:
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Today's Topics:
1. Re: OneWay audio after hold->unhold (Bote Man)
2. Re: Compiling under SmartOS (Stanislav Sinyagin)
3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa)
---------- Forwarded message ----------
From: Bote Man <bote_radio at botecomm.com<mailto:bote_radio at botecomm.com>>
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc:
Date: Mon, 7 Sep 2015 18:03:17 -0400
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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---------- Forwarded message ----------
From: Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc:
Date: Tue, 8 Sep 2015 08:55:37 +0200
Subject: Re: [Freeswitch-users] Compiling under SmartOS
See the update at https://freeswitch.org/jira/browse/FS-7967
I fixed the compilation problems, and now there's a runtime issue.
On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin
<ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
> I see there are some people on the list, working with SmartOS.
>
> The current master fails to compile:
> https://freeswitch.org/jira/browse/FS-7967
>
> Your input will be appreciated.
>
> I just started looking around and getting the feeling what SmartOS is.
> I worked with Solaris quite a lot, but that was almost 10 years ago.
---------- Forwarded message ----------
From: Prashanth Devarajappa <Prashanth.Devarajappa at enghouse.com<mailto:Prashanth.Devarajappa at enghouse.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc:
Date: Tue, 8 Sep 2015 08:53:42 +0000
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold
Hi Bob,
I am told that our Avaya is “avaya communication manager is the 8300”
So far with our testing, we have not seen any issues related to transfer. (May be we have not done extensive transfer tests)
Prashanth
From: Bob Hartwig [mailto:bobjectsfreeswitch at gmail.com<mailto:bobjectsfreeswitch at gmail.com>]
Sent: 07 September 2015 17:42
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
Is this an Avaya AS5300 by any chance? Those have some serious interop issues when it comes to transfers. I had to struggle mightily to get a client's product (not Freeswitch) to interoperate with it. I'm really interested to hear if this gets resolved for Freeswitch.
On Fri, Sep 4, 2015 at 11:09 AM, 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
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
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