Re: [Freeswitch-users] SRTP support
Hi Brian, Thank you! Should I just add this line to vars.xml to "Accept/Offer SAVP/AVP with SAVP preferred"? <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional"/> Tom
Read the very long description on SRTP in the vars.xml file, It has some extensive documentation on this topic. (Hint: were not a proxy.)
On Wed, Oct 26, 2016 at 11:42 AM, Tom Chen <chentom60 at hotmail.com<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
Hello,
By default, Freeswitch seems always translates incoming SRTP packets to outbound RTP only packets as the INVITE SIP message to Receiver from FS contains only RTP/AVP audio media in SDP section even if the Caller's INVITE has RTP/SAVP audio media only.
Caller to FS
Session Initiation Protocol (INVITE)
. . . .
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31
Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
Media Description, name and address (m): audio 7078 *RTP/SAVP* 96 97 98 99 0 8 101 100 102
FS to Receiver
Session Description Protocol
Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 172.17.10.23
Media Description, name and address (m): audio 25206 *RTP/AVP* 0 8 101 13
Is there a way to let FS send SRTP packets to receiver as the receiver can accept SRTP media only?
Tom
______
Yes and I would limit your cipher suites, If you send our full list to a Polycom it has a heart attack and mirrors back the SDP completely and the phone requires a reboot to fix itself. :) On Wed, Oct 26, 2016 at 12:49 PM, Tom Chen <chentom60@hotmail.com> wrote:
Hi Brian,
Thank you! Should I just add this line to vars.xml to "Accept/Offer SAVP/AVP with SAVP preferred"? <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional"/>
Tom
Read the very long description on SRTP in the vars.xml file, It has some extensive documentation on this topic. (Hint: were not a proxy.)
On Wed, Oct 26, 2016 at 11:42 AM, Tom Chen <chentom60 at hotmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
* Hello, *>>>* By default, Freeswitch seems always translates incoming SRTP packets to *>* outbound RTP only packets as the INVITE SIP message to Receiver from FS *>* contains only RTP/AVP audio media in SDP section even if the Caller's *>* INVITE has RTP/SAVP audio media only. *>>>* Caller to FS *>>* Session Initiation Protocol (INVITE) *>>* . . . . *>>* Session Description Protocol *>>* Session Description Protocol Version (v): 0 *>>* Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31 *>>* Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 *>* stat-summary=loss,dup,jitt,TTL voip-metrics *>>* Media Description, name and address (m): audio 7078 *RTP/SAVP* 96 97 98 *>* 99 0 8 101 100 102 *>>>* FS to Receiver *>>* Session Description Protocol *>>* Session Description Protocol Version (v): 0 *>* Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 *>* 172.17.10.23 *>>* Media Description, name and address (m): audio 25206 *RTP/AVP* 0 8 101 *>* 13 *>>>* Is there a way to let FS send SRTP packets to receiver as the receiver can *>* accept SRTP media only? *>>>* Tom *>>>* ______*
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Brian, Thank you for the suggestion! Tom ________________________________ From: Brian West <brian@freeswitch.org> Sent: Wednesday, October 26, 2016 10:50 AM To: Tom Chen Cc: freeswitch-users@lists.freeswitch.org Subject: Re: SRTP support Yes and I would limit your cipher suites, If you send our full list to a Polycom it has a heart attack and mirrors back the SDP completely and the phone requires a reboot to fix itself. :) On Wed, Oct 26, 2016 at 12:49 PM, Tom Chen <chentom60@hotmail.com<mailto:chentom60@hotmail.com>> wrote: Hi Brian, Thank you! Should I just add this line to vars.xml to "Accept/Offer SAVP/AVP with SAVP preferred"? <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional"/> Tom
Read the very long description on SRTP in the vars.xml file, It has some extensive documentation on this topic. (Hint: were not a proxy.)
On Wed, Oct 26, 2016 at 11:42 AM, Tom Chen <chentom60 at hotmail.com<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
Hello,
By default, Freeswitch seems always translates incoming SRTP packets to outbound RTP only packets as the INVITE SIP message to Receiver from FS contains only RTP/AVP audio media in SDP section even if the Caller's INVITE has RTP/SAVP audio media only.
Caller to FS
Session Initiation Protocol (INVITE)
. . . .
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31
Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
Media Description, name and address (m): audio 7078 *RTP/SAVP* 96 97 98 99 0 8 101 100 102
FS to Receiver
Session Description Protocol
Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 172.17.10.23
Media Description, name and address (m): audio 25206 *RTP/AVP* 0 8 101 13
Is there a way to let FS send SRTP packets to receiver as the receiver can accept SRTP media only?
Tom
______
-- Brian West brian@freeswitch.org<mailto:brian@freeswitch.org> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here<https://freeswitch.org/jira>! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
Hello, It looks like rtcp stream is not available by default, how do I configure freeSwitch to send rtcp stream during a voip call? Tom
Hi, you have to enable this options in sip profile: <param name="rtcp-audio-interval-msec" value="5000"/> <param name="rtcp-video-interval-msec" value="5000"/> Freeswitch will add rtcp attribute to SDP. On Fri, Nov 4, 2016 at 7:40 PM, Tom Chen <chentom60@hotmail.com> wrote:
Hello,
It looks like rtcp stream is not available by default, how do I configure freeSwitch to send rtcp stream during a voip call?
Tom
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participants (3)
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Brian West
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Tom Chen
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Volodymyr Fedorov