Hello. I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines: m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both? Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600) SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/> Please, advise. Thanks!
set the global var sdp_m_per_ptime=false in vars.xml 2012/2/21 Mihail <pazuzu@ukr.net>:
Hello.
I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines:
m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported
How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both?
Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)
SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/>
Please, advise. Thanks!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thank you for replay. I tried to set up in ../conf/vars.xml <X-PRE-PROCESS cmd="set" data="sdp_m_per_ptime=false"/> After reloadxml and shutdown. It did not help. An issue still exist. --- Исходное сообщение --- От кого: "Anthony Minessale" <anthony.minessale@gmail.com> Кому: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Дата: 21 февраля 2012, 18:29:21 Тема: Re: [Freeswitch-users] How to disable two 'm=' lines
set the global var sdp_m_per_ptime=false in vars.xml 2012/2/21 Mihail <pazuzu@ukr.net>:
Hello.
I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines:
m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported
How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both?
Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)
SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/>
Please, advise. Thanks!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org> http://www.freeswitchsolutions.com> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com> Official FreeSWITCH Sites http://www.freeswitch.org> http://wiki.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>
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale@gmail.comIRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.orggoogletalk:conf+888@conference.freeswitch.orgpstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:htt...
i missed the bit that it was related to srtp this is a joke that they don't accept both its part of the specs. anyway, Update to HEAD and try this new variable (i had to resist calling it cmon_asterisk_really=true) sdp_secure_savp_only=true 2012/2/22 Mihail <pazuzu@ukr.net>:
Thank you for replay. I tried to set up in ../conf/vars.xml
<X-PRE-PROCESS cmd="set" data="sdp_m_per_ptime=false"/>
After reloadxml and shutdown. It did not help. An issue still exist.
--- Исходное сообщение --- От кого: "Anthony Minessale" <anthony.minessale@gmail.com> Кому: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Дата: 21 февраля 2012, 18:29:21 Тема: Re: [Freeswitch-users] How to disable two 'm=' lines
set the global var sdp_m_per_ptime=false in vars.xml
2012/2/21 Mihail <pazuzu@ukr.net>:
Hello.
I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines:
m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported
How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both?
Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)
SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/>
Please, advise. Thanks!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I have updated: FreeSWITCH Version 1.0.head (git-fbe4e64 2012-02-22 23-08-19 -0600) After that, in vars.xml: <X-PRE-PROCESS cmd="set" data="sdp_secure_savp_only=true"/> in sip_profiles/internal.xml: <param name="sdp_secure_savp_only" value="true"/> in dialplan/public.xml: <action application="bridge" data="{sdp_secure_savp_only=true}sofia/internal/${dialed_extension}%${domain_name}"/> or <action application="export" data="nolocal:sdp_secure_savp_only=true"/> or <action application="set" data="nolocal:sdp_secure_savp_only=true"/> Debug output return: EXECUTE sofia/external/123456789@Y.Y.Y.Y bridge({sdp_secure_savp_only=true}sofia/internal/987654321%X.X.X.X) 2012-02-23 19:47:39.499280 [DEBUG] switch_event.c:1522 Parsing variable [sdp_secure_savp_only]=[true] and two m= lines. v=0 o=FreeSWITCH 1329990231 1329990232 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 29028 RTP/SAVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:u6q91ZUqwrEMnd8/w3YsLijAMYfvCoPc/63vJL5o a=silenceSupp:off - - - - a=ptime:20 m=audio 29028 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 --- Исходное сообщение --- От кого: "Anthony Minessale" <anthony.minessale@gmail.com> Кому: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Дата: 23 февраля 2012, 04:41:46 Тема: Re: [Freeswitch-users] How to disable two 'm=' lines
i missed the bit that it was related to srtp this is a joke that they don't accept both its part of the specs. anyway, Update to HEAD and try this new variable (i had to resist calling it cmon_asterisk_really=true) sdp_secure_savp_only=true 2012/2/22 Mihail <pazuzu@ukr.net>:
Thank you for replay. I tried to set up in ../conf/vars.xml
<X-PRE-PROCESS cmd="set" data="sdp_m_per_ptime=false"/>
After reloadxml and shutdown. It did not help. An issue still exist.
--- Исходное сообщение --- От кого: "Anthony Minessale" <anthony.minessale@gmail.com> Кому: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Дата: 21 февраля 2012, 18:29:21 Тема: Re: [Freeswitch-users] How to disable two 'm=' lines
set the global var sdp_m_per_ptime=false in vars.xml
2012/2/21 Mihail <pazuzu@ukr.net>:
Hello.
I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines:
m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported
How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both?
Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)
SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/>
Please, advise. Thanks!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org>> http://www.freeswitchsolutions.com>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com>> Official FreeSWITCH Sites http://www.freeswitch.org>> http://wiki.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>>
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/> ClueCon http://www.cluecon.com/> Twitter: http://twitter.com/FreeSWITCH_wire> AIM: anthm MSN:anthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org> http://www.freeswitchsolutions.com> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com> Official FreeSWITCH Sites http://www.freeswitch.org> http://wiki.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>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org> http://www.freeswitchsolutions.com> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com> Official FreeSWITCH Sites http://www.freeswitch.org> http://wiki.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>
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale@gmail.comIRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.orggoogletalk:conf+888@conference.freeswitch.orgpstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:htt...
participants (2)
-
Anthony Minessale
-
Mihail