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

[Anthony Minessale]

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
>>
>> 
>> 
>>
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>>
>
>
>
> --
> Anthony Minessale II
>
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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/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900