[Freeswitch-users] How to disable two 'm=' lines

Thu Feb 23 20:55:05 MSK 2012

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 at 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 at gmail.com>
Кому: "FreeSWITCH Users Help" <freeswitch-users at 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 at 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 at gmail.com>
> Кому: "FreeSWITCH Users Help" <freeswitch-users at 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 at 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 at freeswitch.org>> http://www.freeswitchsolutions.com>>
>> 
>> >>
>> Official FreeSWITCH Sites
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/> ClueCon http://www.cluecon.com/> Twitter: http://twitter.com/FreeSWITCH_wire>
> AIM: anthm
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>
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org> http://www.freeswitchsolutions.com>
> 
> >
> Official FreeSWITCH Sites
> http://www.freeswitch.org> http://wiki.freeswitch.org> http://www.cluecon.com>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 at freeswitch.org> http://www.freeswitchsolutions.com>
> 
> >
> Official FreeSWITCH Sites
> http://www.freeswitch.org> http://wiki.freeswitch.org> http://www.cluecon.com>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 at hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale at gmail.comIRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.orggoogletalk:conf+888 at conference.freeswitch.orgpstn:+19193869900

_________________________________________________________________________
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