I use a FreePBX installation with Asterisk 13.14 (and test on 14 also). I want to use multicast paging and play a prerecorded announcement file (for evacuation).
To achieve this I use the announcement option within dial app. see:
exten => 881,n,dial(MulticastRTP/basic/224.100.100.100:5555,10,gFA(${ANNOUNCEMENT})L(2000))
Some phones work fine with this (eg. snom d715 or Yealink T42G). The recorded file and the live page is correct.
But some phones like Yealink T23G or Grandstream gxp21xx don’t play the recorded file with multicast. The sound is totally chopped and not understandable. Codec is defined as PCMU. The live multicast paging after announcement is OK.
The support of Yealink found out that timing isn’t according to RFC standard. I think is similar than an old issue on Multicast RTP. This was for live streaming with page/dial multicast.
Here is the original comment from Yealink support:
Kevin_Yealink Reply Mar 10, 2017 02:31:33 +0800
Hope this e-mail find you well.
Today i check with R&D. Because T42G and T23G use different sound solution from different vendor, so it behavior differently.
R&D said that the RTP don’t follow up the RFC standard. So when T23G receive the RTP file, it will have such issue.
For the issue, can you check on the server side. Before the server send the RTP to the phone, it will convert the sound file to RTP, and it will use the coded and the ptime to calculate sampling*ptime=ts.
For the issue, maybe you can ask someone for help to set ts as 160, R&D said it can solve the issue.
Because T23G and T42G use different sound solution, it is design by this. We can’t change it. Hope you can understand