SIP Connection Audio Quality Correction Inquiry

[sangsoo]

Hello.
I'm developing a web-based SIP service.
----Version info-----
wcs_version=5.3.243-b287c9cf9f9fb400f97398296810c5ca214b5199
wcs_client_version=2.0.269-7ca52229c68b6b769e3e67e29c7e4cf887237822

When connecting via WCS in a web browser, the audio tempo fluctuates between faster and slower, but it sounds quite stable when using a softphone. (Calls using a browser sound normal.)
The callee side is a TTS system. (only Uses G.711)
I found articles about buffer and jitter management. Is there a way to correct this using WCS?

Is there really a difference compared to a softphone?
Please reply. Thank you.

 

[Max]

Good day.
Please clarify the case:
1. Caller establishes audio only call from browser to the TTS
2. Caller hears audio from the TTS sometimes fast, sometimes slow
or
2. Caller hears distorted audio from the TTS (missing phrases, robotic voice etc)?

(Calls using a browser sound normal.)

What do you mean: SIP calls between two browsers via WCS or something else?
Anyway, please exclude audio transcoding by disabling Opus codec for SIP calls and leaving G711 (PCMA and PCMU only)

codecs_exclude_sip=opus,mpeg4-generic,flv,mpv,g729,speex16,g722,vp8

You can also exclude a possible packet loss (which may lead to audio distortion) between browser and WCS by switching to WebRTC over TCP

ice_tcp_transport=true

 

[sangsoo]

Thank you for reply.

To explain in more detail,
1. When a caller makes a call, an Bot answers the call.
2. The Bot analyzes the inquiry and responds in a human-like manner.

The problem is that the Bot's voice can sometimes be audible to the caller, but sometimes it can be distorted.
Typical symptoms include responses that are either too fast or too slow. Speech may also be interrupted mid-sentence.

Also, if the caller uses Softphone instead of the web, the call is normal.

For comparison purposes, opening two phone_min pages and connecting via g.711 works fine.
Currently, it's difficult to apply ice TCP.

Is there a way to fix this with flashphoner.properties settings?
Will this conflict with the settings I'm using?

Thanks.

 

[Max]

Please gather a traffic pcap dump between WCS and PBX TTS.

tcpdump -s 4096 -w log.pcap

Analyze this dump in Wireshark and make sure G.711 RTP packets are captured by the tcpdump.
Once it is captured, please send us using form
We will setup internal test with sipp to replay this dump and reproduce issue.