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@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@10.0.0.57! 2013-11-21 09:35:04.580658 [NOTICE] mod_dptools.c:1225 Channel [sofia/internal/1009@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@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@10.0.0.57) Ended 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close Channel sofia/internal/1009@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