<div dir="ltr">Do you get different results when you call it with some other phone in place of sofia_cli ?<div><br></div><div>Can you get a full capture of the debug logs</div><div><br></div><div>sofia global siptrace on</div>
<div>console loglevel debug</div><div>sofia tracelevel alert</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Nov 21, 2013 at 12:13 PM, Jean-Paul Iribarren <span dir="ltr"><<a href="mailto:jeanpaul.iribarren@free.fr" target="_blank">jeanpaul.iribarren@free.fr</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi FS wizards, first message on this list... Apologies if this post<br>
belongs to another section (please tell me if this is the case).<br>
<br>
First, a big thank you to all members of the team for this amazing piece<br>
of software. Great job here. Now, on for my issue: I am currently<br>
testing a Linux sofsip_cli SIP client on an embedded device, connected<br>
to a Win32 FreeSWITCH version, and I can't get correct RX audio on my<br>
platform. I ran the same test with a linphonec client on the same<br>
embedded device for comparison purposes, and the rx audio is good. Both<br>
clients end up using the same RTP configuration (PCMU/8000). Preliminary<br>
conclusion: my sofsip_cli client is buggy. But there is more...<br>
<br>
I ended up checking data exchanged between endpoints using Wireshark,<br>
and I was surprised to see that in front of sofsip_cli, FreeSWITCH seems<br>
to timestamp TX RTP packets incorrectly: it sends 160 samples/RTP<br>
packet, but the timestamps of consecutive TX packets are increased in<br>
320 units steps, and FS sends 25 RTP packets per second on average.<br>
Consecutive RTP packets actually contain consecutive PCMU data (from the<br>
audio point of view), resulting in a sloooow, chopped rx audio coming<br>
out of the loudspeaker of my device. Using Wireshark embedded player to<br>
play the VoIP audio results in the same slow / chopped audio, which<br>
seems to prove that FS actually sent bad RTP.<br>
<br>
On the other hand, when I use linphonec in the very same conditions, I<br>
can see that FreeSWITCH sends 50 RTP packets per second on average. Each<br>
RTP packet contains 160 samples, and timestamps of consecutive are<br>
increased in 160 units steps, resulting in correct RX audio coming out<br>
of the loudspeaker of my device. FS logs in console window are the same<br>
for both cases, below is what I get for the sofsip_cli case:<br>
<br>
2013-11-21 09:35:00.549253 [NOTICE] switch_channel.c:1055 New Channel<br>
sofia/internal/<a href="mailto:1009@10.0.0.57">1009@10.0.0.57</a> [527f855e-3856-4805-b72c-da59b3a6d292]<br>
2013-11-21 09:35:04.502530 [INFO] mod_dialplan_xml.c:558 Processing 1009<br>
<1009>->1001 in context default<br>
2013-11-21 09:35:04.533781 [INFO] switch_ivr_async.c:3631 Bound B-Leg:<br>
*1 execute_extension::dx XML features<br>
2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg:<br>
*2 record_session::C:/Program<br>
Files/FreeSWITCH/recordings/1009.2013-11-21-09-35-04.wav<br>
2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg:<br>
*3 execute_extension::cf XML features<br>
2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg:<br>
*4 execute_extension::att_xfer XML features<br>
2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot<br>
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]<br>
2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot<br>
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]<br>
2013-11-21 09:35:04.580658 [INFO] mod_dptools.c:3201 Originate Failed.<br>
Cause: USER_NOT_REGISTERED<br>
2013-11-21 09:35:04.580658 [NOTICE] sofia_media.c:92 Pre-Answer<br>
sofia/internal/<a href="mailto:1009@10.0.0.57">1009@10.0.0.57</a>!<br>
2013-11-21 09:35:04.580658 [NOTICE] mod_dptools.c:1225 Channel<br>
[sofia/internal/<a href="mailto:1009@10.0.0.57">1009@10.0.0.57</a>] has been answered<br>
2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel<br>
loopback/app=voicemail:default 10.0.0.57 1001-a<br>
[f2aa95f1-0631-4590-9fa4-c4dc1c96c88e]<br>
2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1053 Rename Channel<br>
loopback/app=voicemail:default 10.0.0.57 1001-a->loopback/voicemail-a<br>
[f2aa95f1-0631-4590-9fa4-c4dc1c96c88e]<br>
2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel<br>
loopback/voicemail-b [fc478cfd-e932-450a-9f50-2a5a76acc907]<br>
2013-11-21 09:35:05.611948 [NOTICE] mod_loopback.c:947 Pre-Answer<br>
loopback/voicemail-a!<br>
2013-11-21 09:35:05.611948 [NOTICE] mod_dptools.c:1260 Pre-Answer<br>
loopback/voicemail-b!<br>
2013-11-21 09:35:11.549676 [NOTICE] sofia.c:715 Hangup<br>
sofia/internal/<a href="mailto:1009@10.0.0.57">1009@10.0.0.57</a> [CS_EXECUTE] [NORMAL_CLEARING]<br>
2013-11-21 09:35:11.565301 [NOTICE] switch_ivr_bridge.c:733 Hangup<br>
loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL]<br>
2013-11-21 09:35:11.565301 [NOTICE] mod_loopback.c:553 Hangup<br>
loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL]<br>
2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1598 Session 5<br>
(loopback/voicemail-a) Ended<br>
2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1602 Close<br>
Channel loopback/voicemail-a [CS_DESTROY]<br>
2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 4<br>
(sofia/internal/<a href="mailto:1009@10.0.0.57">1009@10.0.0.57</a>) Ended<br>
2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close<br>
Channel sofia/internal/<a href="mailto:1009@10.0.0.57">1009@10.0.0.57</a> [CS_DESTROY]<br>
2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 6<br>
(loopback/voicemail-b) Ended<br>
2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close<br>
Channel loopback/voicemail-b [CS_DESTROY]<br>
<br>
I ran my tests by calling an unregistered extension (1001), therefore<br>
resulting in FS sending voice messaging audio, and hanging up on client<br>
after a few seconds. I have compared the SIP handshakes for both<br>
clients, and I can't see any significant difference (except that<br>
linphonec supports several codecs, while sofsip_cli only supports<br>
G.711/PCMU).<br>
<br>
Regarding FS config (this may or may not be relevant), I had to tweak it<br>
a bit to support sofsip_cli. I added the two following lines in the<br>
dialplan for the relevant extensions (file default.xml):<br>
<br>
<action application="set" data="disable_rtp_auto_adjust=true"/><br>
<action application="set" data="zrtp_secure_media=false"/><br>
<br>
I also changed the two following lines to add video codecs in file vars.xml:<br>
<br>
<X-PRE-PROCESS cmd="set"<br>
data="global_codec_prefs=G722,PCMU,PCMA,GSM,H263,H263-1998,H264"/><br>
<X-PRE-PROCESS cmd="set"<br>
data="outbound_codec_prefs=PCMU,PCMA,GSM,H263,H263-1998,H264"/><br>
<br>
The rest of the config is plain vanilla.<br>
<br>
The (small) Wireshark traces of both tests can be downloaded from<br>
Box:<br>
<br>
- linphonec trace (FS TX RTP ok): file linphone.dac.pcap is available at<br>
<a href="https://app.box.com/s/63p8ayrr3dc6w1d9wm33" target="_blank">https://app.box.com/s/63p8ayrr3dc6w1d9wm33</a><br>
<br>
- sofsip_cli trace (FS TX RTP ko): file ssc.pcap is available at<br>
<a href="https://app.box.com/s/z1cde37xg5teh9d690oy" target="_blank">https://app.box.com/s/z1cde37xg5teh9d690oy</a><br>
<br>
My device has IP address 10.0.0.186, and FS is installed on my PC<br>
(running Win XP Pro) at 10.0.0.57.<br>
<br>
My FreeSWITCH version is "1.5.6b+git~20131101T202135Z~2589bf7750~32bit<br>
(git 2589bf7 2013-11-01 20:21:35Z 32bit)"<br>
<br>
Sorry for the long post (I tried to give all relevant info), and thank<br>
you in advance for your comments and help on this weird issue.<br>
--<br>
JPI<br>
<br>
<br>
_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-dev mailing list<br>
<a href="mailto:FreeSWITCH-dev@lists.freeswitch.org">FreeSWITCH-dev@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-dev" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-dev</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
Twitter: <a href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:+19193869900
</div>