[Freeswitch-dev] FreeSWITCH erroneous RTP timestamps in front of sofsip_cli client?

Fri Nov 22 00:06:23 MSK 2013

Do you get different results when you call it with some other phone in
place of sofia_cli ?

Can you get a full capture of the debug logs

sofia global siptrace on
console loglevel debug
sofia tracelevel alert

On Thu, Nov 21, 2013 at 12:13 PM, Jean-Paul Iribarren <
jeanpaul.iribarren at free.fr> wrote:

> Hi FS wizards, first message on this list... Apologies if this post
> belongs to another section (please tell me if this is the case).
>
> First, a big thank you to all members of the team for this amazing piece
> of software. Great job here. Now, on for my issue: I am currently
> testing a Linux sofsip_cli SIP client on an embedded device, connected
> to a Win32 FreeSWITCH version, and I can't get correct RX audio on my
> platform. I ran the same test with a linphonec client on the same
> embedded device for comparison purposes, and the rx audio is good. Both
> clients end up using the same RTP configuration (PCMU/8000). Preliminary
> conclusion: my sofsip_cli client is buggy. But there is more...
>
> I ended up checking data exchanged between endpoints using Wireshark,
> and I was surprised to see that in front of sofsip_cli, FreeSWITCH seems
> to timestamp TX RTP packets incorrectly: it sends 160 samples/RTP
> packet, but the timestamps of consecutive TX packets are increased in
> 320 units steps, and FS sends 25 RTP packets per second on average.
> Consecutive RTP packets actually contain consecutive PCMU data (from the
> audio point of view), resulting in a sloooow, chopped rx audio coming
> out of the loudspeaker of my device. Using Wireshark embedded player to
> play the VoIP audio results in the same slow / chopped audio, which
> seems to prove that FS actually sent bad RTP.
>
> On the other hand, when I use linphonec in the very same conditions, I
> can see that FreeSWITCH sends 50 RTP packets per second on average. Each
> RTP packet contains 160 samples, and timestamps of consecutive are
> increased in 160 units steps, resulting in correct RX audio coming out
> of the loudspeaker of my device. FS logs in console window are the same
> for both cases, below is what I get for the sofsip_cli case:
>
> 2013-11-21 09:35:00.549253 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/1009 at 10.0.0.57 [527f855e-3856-4805-b72c-da59b3a6d292]
> 2013-11-21 09:35:04.502530 [INFO] mod_dialplan_xml.c:558 Processing 1009
> <1009>->1001 in context default
> 2013-11-21 09:35:04.533781 [INFO] switch_ivr_async.c:3631 Bound B-Leg:
> *1 execute_extension::dx XML features
> 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg:
> *2 record_session::C:/Program
> Files/FreeSWITCH/recordings/1009.2013-11-21-09-35-04.wav
> 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg:
> *3 execute_extension::cf XML features
> 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg:
> *4 execute_extension::att_xfer XML features
> 2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2013-11-21 09:35:04.580658 [INFO] mod_dptools.c:3201 Originate Failed.
> Cause: USER_NOT_REGISTERED
> 2013-11-21 09:35:04.580658 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1009 at 10.0.0.57!
> 2013-11-21 09:35:04.580658 [NOTICE] mod_dptools.c:1225 Channel
> [sofia/internal/1009 at 10.0.0.57] has been answered
> 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel
> loopback/app=voicemail:default 10.0.0.57 1001-a
> [f2aa95f1-0631-4590-9fa4-c4dc1c96c88e]
> 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1053 Rename Channel
> loopback/app=voicemail:default 10.0.0.57 1001-a->loopback/voicemail-a
> [f2aa95f1-0631-4590-9fa4-c4dc1c96c88e]
> 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel
> loopback/voicemail-b [fc478cfd-e932-450a-9f50-2a5a76acc907]
> 2013-11-21 09:35:05.611948 [NOTICE] mod_loopback.c:947 Pre-Answer
> loopback/voicemail-a!
> 2013-11-21 09:35:05.611948 [NOTICE] mod_dptools.c:1260 Pre-Answer
> loopback/voicemail-b!
> 2013-11-21 09:35:11.549676 [NOTICE] sofia.c:715 Hangup
> sofia/internal/1009 at 10.0.0.57 [CS_EXECUTE] [NORMAL_CLEARING]
> 2013-11-21 09:35:11.565301 [NOTICE] switch_ivr_bridge.c:733 Hangup
> loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL]
> 2013-11-21 09:35:11.565301 [NOTICE] mod_loopback.c:553 Hangup
> loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL]
> 2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1598 Session 5
> (loopback/voicemail-a) Ended
> 2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1602 Close
> Channel loopback/voicemail-a [CS_DESTROY]
> 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 4
> (sofia/internal/1009 at 10.0.0.57) Ended
> 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close
> Channel sofia/internal/1009 at 10.0.0.57 [CS_DESTROY]
> 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 6
> (loopback/voicemail-b) Ended
> 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close
> Channel loopback/voicemail-b [CS_DESTROY]
>
> I ran my tests by calling an unregistered extension (1001), therefore
> resulting in FS sending voice messaging audio, and hanging up on client
> after a few seconds. I have compared the SIP handshakes for both
> clients, and I can't see any significant difference (except that
> linphonec supports several codecs, while sofsip_cli only supports
> G.711/PCMU).
>
> Regarding FS config (this may or may not be relevant), I had to tweak it
> a bit to support sofsip_cli. I added the two following lines in the
> dialplan for the relevant extensions (file default.xml):
>
>      <action application="set" data="disable_rtp_auto_adjust=true"/>
>      <action application="set" data="zrtp_secure_media=false"/>
>
> I also changed the two following lines to add video codecs in file
> vars.xml:
>
>    <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G722,PCMU,PCMA,GSM,H263,H263-1998,H264"/>
>    <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMU,PCMA,GSM,H263,H263-1998,H264"/>
>
> The rest of the config is plain vanilla.
>
> The (small) Wireshark traces of both tests can be downloaded from
> Box:
>
> - linphonec trace (FS TX RTP ok): file linphone.dac.pcap is available at
> https://app.box.com/s/63p8ayrr3dc6w1d9wm33
>
> - sofsip_cli trace (FS TX RTP ko): file ssc.pcap is available at
> https://app.box.com/s/z1cde37xg5teh9d690oy
>
> My device has IP address 10.0.0.186, and FS is installed on my PC
> (running Win XP Pro) at 10.0.0.57.
>
> My FreeSWITCH version is "1.5.6b+git~20131101T202135Z~2589bf7750~32bit
> (git 2589bf7 2013-11-01 20:21:35Z 32bit)"
>
> Sorry for the long post (I tried to give all relevant info), and thank
> you in advance for your comments and help on this weird issue.
> --
> JPI
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Anthony Minessale II

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