[asterisk-ss7] chan_ss7.c:1889 ss7_write: Write buffer full on CIC=1 (wrote only 0 of 160), audio lost.

Jacob Tinning tinning at sifira.dk
Thu May 11 00:12:14 MST 2006


On Wed, 10 May 2006, Mr.Surender Reddy wrote:

> Can anyone help me out to solve the chan_ss7.c:1889 ss7_write:
> Write buffer full on CIC=1 (wrote only 0 of 160), audio lost.We have aserisk
> server with 2 E1 and 2 SS7 signaling running on the same box,We are using
> Digium cards and using chan ss7 solution everything is fine and we get this
> problem very offten.

Are you using the Asterisk / chan_ss7 as a gateway to IP-telephones or
maybe through IAX to another asterisk ?

The "write-buffer full" problem often occur because of jitter on the IP-net.

In IP-networks, there is allways a time-delay from the time a packet is
sent to it get received at another host. This delay is unfortunately not
allways the same. Sometimes, the packets will arrive too slow, and
sometimes they will arrive too fast. When they arrive too fast, the
send-buffer is filled, chan_ss7 writes the "Write buffer full" and then
drop the packets.

This problem does not occur in the conventional circuit-switched telephone
network, since it is completely synchronous and a dedicated (logical)
circuit is reserved between the sender and receiver.

To get around this problem, you can setup trafic-shaping on the network,
insert a bigger buffer in chan_ss7 (which will cause more delay), to
even-out the difference in time-delay. Maybe it will helpl to configure the
routers to "chop up" big packets in order to get the small RTP packets
going smoothly.

Read more at: http://www.0xdecafbad.com/What-is-Jitter.html
and generally about QoS on IP-networks at
http://www.voip-info.org/wiki/view/QoS

Mvh. Jacob

-- 
Jacob Tinning
System Developer                                           SIFIRA A/S


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