What is the difference between DAHDI Trunk and SIP Trunk?


(Alex Knee) #1

Hi all
What is the difference between DAHDI Trunk and SIP Trunk?
Please advise which one should I use?

Thanks


(David55) #2

You really need to do a lot of background reading on telephony systems, but:

DAHDI trunks are circuit switched, which means resources are dedicated to the call for the whole duration (e.g. as separate wires, for analogue, or as a particular few microseconds every 125 microseconds, on a single cable, for digital), whereas SIP is packet switched, which means that the conversation is broken up into chunks and each chunk has to wait for resources to be available to reach the next hop.

With DAHDI trunks, you will either get congestion immediately, or you will get a good connection for the whole call. With SIP trunks the connection quality can be variable, including temporary drop outs, depending on how overloaded the network is, but the call will tend to get set up eve if it is bad. You will be competing with other sorts of traffic, e.g. streaming movies, and file downloads.

With DAHDI trunks, you will generally get a maximum of 125 microseconds delay, in addition to speed of light considerations, for each stage of switching. On SP with standard settings, you will get a minimum of 20ms end to end delay (40ms round trip), in addition to speed of light considerations, but with overloaded networks, this can become 100s of milliseconds. This means that echoes are obvious on SIP, although SIP equipment will try to predict and remove the echo.

WIth DAHDI, you don’t get to choose the codec, which is generally one for 300 to 3.4kHz, which is communications speech quality, but not hi-fi, although it will represent music reasonably well. This is the same as the PSTN backbone, so you can’t improve on it for PSTN calls. With SIP you may get to choose a range of codecs, some are designed to reduce data rate, typically at the cost of being speech only, with music badly mangled, and some designed for higher quality audio, but PSTN calls cannot take advantage of this.

DAHDI represents older technology in the PSTN backbone. They then moved to a specialised packet switching system, ATM, which is designed to avoid some of the problems mentioned above, by reserving bandwidth, and keeping delays constant and low. The backbone is now moving to using IP networks, because voice has become a minor part of their use, and it is possible to reserve bandwidth on them for voice without having much effect on other traffic.

DAHDI trunks require extra hardware, require a separate physical connection to the PSTN, and, I suspect are more expensive than SIP ones. SIP ones can share existing internal and external data network hardware, and you may already have spare capacity on that.

However, if you are asking that question, you really need to do a lot more background reading than you can get from forum replies.

Note that analogue trunks are not are not a good choice where there is any automation of calls, but circuit switched digital trunks are probably how your SIP provider would connect to the PSTN, Both would be implemented with DAHDI, but they are very different. I’d advise against analogue trunks if you don’t already have them.


(Matt Schubeck) #3

David55 is really thorough with his explanation and analysis, but maybe it’s too much for someone that really does need to learn a bit more about VoIP telephony than what you currently know. Therefore, I’ll try to simplify it just a bit:

DAHDI is generally only used if you have physical trunks you’re going to connect - a T1/PRI, or analog POTS lines, etc. You would order these trunks from your PSTN provider, and you set them up in FreePBX as DAHDI trunks after installing the corresponding hardware.

SIP is network-based and can be ordered from hundreds of different providers without requiring any specific hardware to be installed.

Therefore, what you should use is relative to your installation scenario:

  • If you’re creating a new telephony system from scratch and aren’t sure what trunks to order, it will typically be better and more cost-efficient to go with SIP trunks.

  • If you’re integrating a new FreePBX installation into an existing system, or replacing an existing conventional PBX and don’t want to mess around with porting numbers and ironing out configurations, then keeping your existing trunks and setting them up as DAHDI in FreePBX is the way to go.

  • PSTN trunks (DAHDI in FreePBX) are generally more reliable than SIP trunks (not automatically so, there are exceptions on both sides), but offer almost no realistic redundancy in case of failure.

  • SIP trunks easily allow for redundancy, which for most people should make up for the (slightly) diminished reliability.


(system) closed #4

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