<div> </div>
<div>Hello</div>
<div>I have tried the following:</div>
<div>[jitter]<br>jbenable = yes<br>jbmaxsize = 1000<br>jbresyncthreshold = 1000<br>jbimpl = adaptive<br></div>
<div>I am getting bad / chopped sound and error messages:</div>
<div>[2011-02-08 22:09:06] WARNING[31505]: abstract_jb.c:468 create_jb: Failed to put first frame in the jitterbuffer on channel 'SS7/siuc/30'<br>[2011-02-08 22:09:06] WARNING[31505]: chan_iax2.c:1011 jb_warning_output: Resyncing the jb. last_delay 0, this delay -247066746, threshold 1000, new offset 247066746<br>
[2011-02-08 22:09:06] WARNING[31505]: chan_iax2.c:1011 jb_warning_output: Resyncing the jb. last_delay 0, this delay -247066739, threshold 1000, new offset 247066739<br>[2011-02-08 22:09:08] WARNING[31505]: abstract_jb.c:428 jb_get_and_deliver: JB_IMPL_NOFRAME is returned from the adaptive jb when now=2563 >= next=2563, jbnext=2563!<br>
[2011-02-08 22:09:08] WARNING[31505]: abstract_jb.c:428 jb_get_and_deliver: JB_IMPL_NOFRAME is returned from the adaptive jb when now=2564 >= next=2563, jbnext=2563!<br>[2011-02-08 22:09:08] WARNING[31505]: abstract_jb.c:428 jb_get_and_deliver: JB_IMPL_NOFRAME is returned from the adaptive jb when now=2565 >= next=2563, jbnext=2563!<br>
</div>
<div>Note that though chan_iax2.c appears, I am not using IAX2 - just plain SIP:</div>
<div>SIP -> asterisk --> ss7</div>
<div> </div>
<div>After a few seconds (but any normal cust would have hung up by then), the sound quality improves and then goes...</div>
<div> </div>
<div>I've changed the buffer to fixed, but the 1000ms gives a clear 1 sec delay between the sip and the ISDN. I've reduced it to 160ms, and the sound quality is fine. I need now to see if this solves the write buffer issue. </div>
<div> </div>
<div> </div>
<div>Any help on the adaptive mode setting would be welcomed, </div>
<div> </div>
<div>Regards, </div>
<div> </div>
<div>J.<br></div>
<div class="gmail_quote">On Tue, Feb 8, 2011 at 3:35 PM, Jean Cérien <span dir="ltr"><<a href="mailto:cerien.jean@gmail.com">cerien.jean@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div> </div>
<div>Hello all chan_ss7 users</div>
<div> </div>
<div>I am facing this issue as well, using chan_ss7 1.4.3 and dahdi complete 2.3.0</div>
<div> </div>
<div>There seems to be a consensus to set the jitter buffer max size to 1000ms - I am concerned as will this induce a 1 sec delay ? Or will the adaptative mode allow to optimize this ?</div>
<div> </div>
<div>Is anyone using this parameters in production ? What feedback can you give ?</div>
<div> </div>
<div>I am a bit cautious as this is happening in prod (not test), and dont want to sc*w things up !</div>
<div> </div>
<div>Regards, </div>
<div> </div><font color="#888888">
<div>J.<br><br></div></font>
<div>
<div></div>
<div class="h5">
<div class="gmail_quote">On Tue, Jan 26, 2010 at 5:45 PM, marek cervenka <span dir="ltr"><<a href="mailto:cervajs@fpf.slu.cz" target="_blank">cervajs@fpf.slu.cz</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>> [Jan 26 19:40:39] NOTICE[9960]: l4isup.c:2434 ss7_write: Write buffer full<br>> on CIC=122 (wrote only 0 of 160), audio lost (suppress 17).<br>> [Jan 26 19:40:50] NOTICE[9960]: l4isup.c:2434 ss7_write: Write buffer full<br>
> on CIC=122 (wrote only 0 of 160), audio lost (suppress 0).<br>> [Jan 26 19:41:02] NOTICE[9960]: l4isup.c:2434 ss7_write: Write buffer full<br>> on CIC=122 (wrote only 0 of 160), audio lost (suppress 5).<br>> [Jan 26 19:41:20] NOTICE[9960]: l4isup.c:2434 ss7_write: Write buffer full<br>
> on CIC=122 (wrote only 0 of 160), audio lost (suppress 7).<br>> [Jan 26 19:41:38] NOTICE[9960]: l4isup.c:2434 ss7_write: Write buffer full<br>> on CIC=122 (wrote only 0 of 160), audio lost (suppress 10).<br><br>
</div>from chan_ss7 faq:<br><br>===text====<br>Write buffer full on CIC=4 (wrote only 0 of 160), audio lost.<br><br>[Sep 19 18:41:08] NOTICE[10930] l4isup.c: Write buffer full on CIC=4<br>(wrote only 0 of 160), audio lost.<br>
[Sep 19 18:41:08] NOTICE[10930] l4isup.c: Write buffer full on CIC=4<br>(wrote only 0 of 160), audio lost.<br>[Sep 19 18:41:08] NOTICE[10930] l4isup.c: Write buffer full on CIC=4<br>(wrote only 0 of 160), audio lost.<br><br>
Actually, the "Write buffer" is a kind of jitter-buffer. It is inside the<br>zaptel driver and is initialized by chan_ss7 to 4 * 160 bytes (80ms). You<br>can increase it to some other multiple of 160, but it requires a<br>
recompile, and will also increase the audio-delay.<br><br>The "write-buffer full" problem often occur because of jitter on the<br>IP-net.<br><br>In IP-networks, there is allways a time-delay from the time a packet is<br>
sent to it get received at another host. This delay is unfortunately not<br>allways the same. Sometimes, the packets will arrive too slow, and<br>sometimes they will arrive too fast. When they arrive too fast, the<br>send-buffer is filled, chan_ss7 writes the "Write buffer full" and then<br>
drop the packets.<br><br>This problem does not occur in the conventional circuit-switched telephone<br>network, since it is completely synchronous and a dedicated (logical)<br>circuit is reserved between the sender and receiver.<br>
<br>To get around this problem, you can setup trafic-shaping on the network,<br>insert a bigger buffer in chan_ss7 (which will cause more delay), to<br>even-out the difference in time-delay. Maybe it will helpl to configure<br>
the routers to "chop up" big packets in order to get the small RTP packets<br>going smoothly.<br><br>===end text===<br><br>you can try:<br>- newer version of chan_ss7<br>- jitter buffer option in ss7.conf<br>[jitter]<br>
jbenable = yes<br>jbmaxsize = 1000<br>jbresyncthreshold = 1000<br>jbimpl = adaptive<br><br><br>---------------------------------------<br><font color="#888888">Marek Cervenka<br>=======================================<br>
</font>
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