Hi, I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? This is causing problems with the Sipura SPA2000 on our network. The SPA does not like this and is treating many packets as lost packets (even though an Ethereal RTP Analysis trace shows they were not lost). Regards, Andres http://www.telesip.net
> I have a question regarding the Asterisk Packet Time for SIP Calls. It is > hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that > these packets are not spaced out at 20ms. In general you see something like: > > Packet 50 - Delay 50ms > Packet 51 - Delay 5ms > Packet 52 - Delay 5ms > Packet 53 - Delay 50ms > Packet 54 - Delay 5ms > Packet 55 - Delay 5ms > > Is there anyway to space them out evenly at 20ms??The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * --> C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Rich
I might be wrong, but isn't is just saying that the packet has been delayed x-ms? I'm not sure it's saying that Packet 52 arrived 5ms after packet 51. Although even if it was, that doesn't mean that it was sent 5ms after packet 51 either. -----Original Message----- From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] Sent: Monday, December 22, 2003 3:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)> > Packet 50 - Delay 50ms > > Packet 51 - Delay 5ms > > Packet 52 - Delay 5ms > > Packet 53 - Delay 50ms > > Packet 54 - Delay 5ms > > Packet 55 - Delay 5ms> The 20 ms is not the inter-packet timing, its the relative content of > what's within the packet. In other words, the packet contains 20ms of > encoded voice.If that is the case, then what is in packet 52 and 55? There's not enough time between packets for 20ms of voice, unless it's repeating audio in the packets... Regards, Andrew _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
The problem occurs when the software is expecting the packet in a certain timeframe so that it can reassemble it in a timely manner. It's not a big deal with a web page or something along that lines. But when a voice application cannot get reassembled in a timely manner, you'll surely notice it! -----Original Message----- From: Joel Maslak To: asterisk-users@lists.digium.com Sent: 12/23/2003 10:41 AM Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) On Tue, 23 Dec 2003, Rich Adamson wrote:> If a collision or dropped packet occurs (in a voip udp environment)there> is no way to retransmit the missing/damaged packet. Missing one packetisn't> a big deal, but if you have collisions and/or dropped packets, thereis a> very high probability that lots of packets will be dropped. If toomany> are dropped, you'll hear the result in the undecoded voice as choppy > voice.Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 hardware automatically resends packets involved in a collision - layer 3 is never aware of it happening (although it may cause additional delay). Eventually the ethernet card will give up if too many collisions occur during retries, but this is very rare in practice unless the network is *VERY* loaded.> Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg > ethernet would handle roughly 20-25 rtp sessions before bumping intothe> problem (your milage may vary). The majority of the folks on this list > seem to be running home/soho systems and would likely never run intothe> issue. But the heavier users will.For a duplex mismatch, my experience is that if one end on a 100 Mb/sec link is half and the other is full, bandwidth is limited to about 8 Mb/sec max. This is based on some tests I've accidentally conducted. If you try to send 9 Mb/sec over that link, yes, some packets will get dropped as they simply won't fit. (But I do agree that for a half-half link, you can get about 20 Mb/sec) -- Joel _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users