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!
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