04-26-2011 08:08 AM - edited 03-21-2019 04:00 AM
Has anyone experienced complaints in regards to the quality of the Music on Hold over a SIP trunk? We've had 2 seperate clients voice concerns
about the quality of their hold music. We've tried both of the .wav files, as well as different tape decks, CD players, and MP3 devices, all with the same results. The codec we've been using is G711. Any advice towards this issue would be great!
Solved! Go to Solution.
04-26-2011 02:47 PM
Hi Andrew,
This is a common problem and also a misunderstood one by many, even myself did not quite get this issue for a long time and become somewhat frustrated by it.
The fact is that audio streaming was never meant for SIP trunks, there are far too many variables that can interfere with the audio quality and these are amplified to a greater degree over just simple voice packets.
In some cases this could be improved by changing the packetization to 40ms rather than 20ms, but then this would only make it work slightly better, it only takes one dropped packet or a bout of latency to ruin the audio stream, and it can take up to 3 times as long to recover from it as opposed to voice packets.
You will find the best MoH to be used on SIP trunks is one with more talking such as advertisements or promo talks, they fair out far better then a constant stream of music, you get far fewer white noise effect or rapid decreases in volume with just plain speak.
I know this does not help you resolve the problem, but I hope it sheds some light on it for you.
(PS) Try and use G.729r8 as your Codec, but if you want a far better result then you need to see if your ITSP can support G.722 wideband as this will provide you the best audio streaming results over any other Codec (Tested and verified this claim more than once).
Cheers,
David.
04-26-2011 02:47 PM
Hi Andrew,
This is a common problem and also a misunderstood one by many, even myself did not quite get this issue for a long time and become somewhat frustrated by it.
The fact is that audio streaming was never meant for SIP trunks, there are far too many variables that can interfere with the audio quality and these are amplified to a greater degree over just simple voice packets.
In some cases this could be improved by changing the packetization to 40ms rather than 20ms, but then this would only make it work slightly better, it only takes one dropped packet or a bout of latency to ruin the audio stream, and it can take up to 3 times as long to recover from it as opposed to voice packets.
You will find the best MoH to be used on SIP trunks is one with more talking such as advertisements or promo talks, they fair out far better then a constant stream of music, you get far fewer white noise effect or rapid decreases in volume with just plain speak.
I know this does not help you resolve the problem, but I hope it sheds some light on it for you.
(PS) Try and use G.729r8 as your Codec, but if you want a far better result then you need to see if your ITSP can support G.722 wideband as this will provide you the best audio streaming results over any other Codec (Tested and verified this claim more than once).
Cheers,
David.
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