RTP timestamps = ptime x 8, so: PTIME:timestamp increment 10:80 20:160 30:240 40:320 50:400 Your endpoints seem to be writing their RTP timestamps correctly. For the endpoint that's all over the place, check the hardware. I've seen bad DSPs, etc cause strange RTP skew before. On Tue, Dec 4, 2012 at 3:40 PM, Ben <ben122uk@gmail.com> wrote:
Hi Brian,
I looked at the timestamps for both RTP streams that are being sent by each client, it appears that the timestamp increments by a value of 400 for each packet, for both streams. I can also see this in FS for each stream -
2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:50:8000]/[G729:18:8000:50:8000] 2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/1002@192.168.17.154 G729/8000 50 ms 400 samples 8000 bits
Looking at the Freeswitch log, I'm guessing that the timestamps set by the client correlate to the sample rate set by Freeswitch? Or am I not on the right track with what you were suggesting?
I've also noticed that when using 50ms ptimes, one of the RTP streams from a client is not very 'steady'. One stream seems acceptable, with only one or two packets straying to 60ms, and jitter hovers around 2-3ms. However the other stream really struggles, the delta(ms) values in wireshark are all over the place. This is the summary of a 20 second RTP stream in wireshark -
Max delta = 120.95 ms at packet no. 1250 Max jitter = 22.00 ms. Mean jitter = 18.12 ms. Max skew = 65.24 ms.
This seems odd since the test I did was all internal on the same LAN. I'll try and track down the cause tomorrow by switching around some kit so that it cuts out any potential problems.
Are there tolerance levels in Freeswitch for RTP streams that are affected during transit? If so, how does Freeswitch react when one of these tolerance levels is exceeded? My FS server seems to just hang during call setup, and no RTP stream is sent out the server to the clients.
Thanks for the help, I'll let you know how I get on tomorrow with some further tests.
Regards,
Ben
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: 04 December 2012 15:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
PCAP's will show you.... I suspect something is telling a lie about the timestamps. Very common. -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543
On Dec 3, 2012, at 9:31 AM, Ben <ben122uk@gmail.com> wrote:
I tried setting the {passthru_ptime_mismatch=true} variable in the dialplan, but no joy unfortunately. I'm also fairly sure that the codecs are matched on both call legs, so shouldn't be having that sort of issue.
I'm going to gather up some evidence and console logs/traces etc so that I can put something together for a bug report on Jira. Is there anything else that someone can suggest I try before it goes on Jira? Is anyone able to re-produce the same problem? Admittedly, I'm not on the latest GIT, but I'm on a recent (last couple of months) stable release.
Thanks in advance for any assistance!
Regards,
Ben
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