OneWay audio after hold->unhold
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
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@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@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@botecomm.com] Sent: 05 September 2015 04:50 To: FreeSWITCH Users Help <freeswitch-users@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@enghouse.com<mailto:Prashanth.Devarajappa@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@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 J Prashanth From: Bote Man [mailto:bote_radio@botecomm.com] Sent: 05 September 2015 04:50 To: FreeSWITCH Users Help <freeswitch-users@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@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@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@botecomm.com] Sent: 07 September 2015 23:03 To: 'FreeSWITCH Users Help' <freeswitch-users@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@botecomm.com] Sent: 05 September 2015 04:50 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@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@enghouse.com<mailto:Prashanth.Devarajappa@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@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@gmail.com] Sent: 07 September 2015 17:42 To: FreeSWITCH Users Help <freeswitch-users@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@enghouse.com<mailto:Prashanth.Devarajappa@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@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
-
Bob Hartwig
-
Bote Man
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Prashanth Devarajappa