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October 2018

59 participants
72 discussions

Video Upgrade/Downgrade and Transcoding
by Carsten Bock 01 Oct '18

01 Oct '18

Hi, I am using FreeSwitch Release: FreeSWITCH (Version 1.9.0 -n20180903T123919Z-1~jessie+1git 5dd4451 2018-08-31 19:05:39Z 64bit) is ready I have the following, rather simple scenario, where I can't figure out, how to configure it properly. I use FreeSwitch as Transcoding B2B-UA/SBC. All the logic is done on a Kamailio in Front of FreeSwitch. Thus my dialplan is rather simple: <context name="public"> <extension name="other"> <condition> <action application="bridge" data="sofia/internal/${destination_number}@${loadbalancer}"/> </condition> </extension> </context> In order to provide transcoding, I have set the following in my internal.xml profile: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="disable-transcoding" value="false"/> <param name="inherit_codec" value="false"/>  <param name="inbound-codec-negotiation" value="generous"/>  <param name="inbound-late-negotiation" value="true"/> In vars.xml, I have the following: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/> <X-PRE-PROCESS cmd="set" data="rtp_pass_codecs_on_stream_change=true"/> <X-PRE-PROCESS cmd="set" data="sip_renegotiate_codec_on_reinvite=true"/> This works like a charm for all Voice-Calls, the B-Party gets an offer including all the codecs from the list, e.g. A-Party supports only G722, B-Party supports only PCMU, so FreeSwitch does the transcoding. However, it leads to issues, when it comes to Video. As soon as I add H264 and VP8 to the codecs list (global_codec_pref / outbound_codec_prefs), the call to my B-Party includes the offer for Video as well, even if the A-Party did not offer Video initially. Is there an easy way to solve this? What are the right settings for such scenario? I tried the following in my dialplan:  <condition field="${switch_r_sdp}" expression="/(.*)(m=video \d+)(.*)/s" break="never"> <anti-action application="export" data="nolocal:absolute_codec_string=G722,PCMA,PCMU"/> </condition> This would remove the Video from the initial invite, if the A-Party did not offer Video initially. However, if I then enable Video at a later stage (Re-INVITE with m=video), FreeSwitch will not forward the new stream to the B-Party and then instead just send a re-INVITE with Audio only to the B-Party. Where did I go wrong? Any ideas? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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RTP problems when using "soft" timer
by Antonio Modesto 01 Oct '18

01 Oct '18

Hi everyone, I have had some problems lately with poor call quality when using FXO channels (mod_khomp) talking to sip phones. Using wireshark I noticed that the RTP stream sent from FS to my SIP device had some packets with wrong timestamps. After changing the rtp timer from soft to posix, the problem was solved. Do you guys know if there is any side effect in using the posix timer? *I am running FS 1.6.20 on a XenServer VM with Debian (Kernel 3.16.0). I can't upgrade to FS 1.8 because Khomp still does not support it. -- Antônio Modesto

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