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

[Mihail]

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!
>
> _________________________________________________________________________
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> 
> >
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-- 
Anthony Minessale II

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