Hi
We have this situation with poor audio quality (noise, echo) and after investigating several of our endpoints we did not find any issue in the network or on the SIP end.
I've sent some logs to logs@flashphoner.com address.
1. This is the call with problems, found in flashphoner_manager.log.2017-09-26-13 at 13:55:
13:55:33,577 INFO agerRemoteRmiService - RMI TCP Connection(675)-172.31.25.198 SEND REST OBJECT ==>
URL:http://localhost:9091/EchoApp/OnCallEvent
OBJECT:
{
"nodeId" : "XQYpQm0VT8TtFA077nCnME2gYn7xDfKj@172.31.25.198 ",
"appKey" : "defaultApp",
"sessionId" : "/39.x.x.x:61315/172.31.25.198:8443",
"callId" : "421e9d1f0d15200050c17ca81984e5bf@172.31.25.199",
"incoming" : true,
"status" : "PENDING",
"caller" : "xxx0297",
"callee" : "192995",
"hasAudio" : true,
"hasVideo" : false,
"sdp" : "v=0\r\no=root 1090804450 1090804450 IN IP4 172.31.25.199\r\ns=Asterisk PBX 1.6.0.26\r\nc=IN IP4 172.31.25.199\r\nt=0 0\r\nm=audio 10956 RTP/AVP 0 8\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=ptime:20\r\na=sendrecv\r\n",
"visibleName" : "xxx0297_24435_12484_7668_1_1506401724.9370701_",
"mediaProvider" : "WebRTC",
"isMsrp" : false,
"holdForTransfer" : false
}
2. Also there is a .pcap with this exact call. (its the stream between Flashphoner server and Asterisk PBX). The RTP stream seems to have Jitter..but I don't know if this Jitter is due to the connection between Client browser and WCS.
3. Version we are using is : FlashphonerWebCallServer-5.0.1994
Any help is highly appreciated.
We have this situation with poor audio quality (noise, echo) and after investigating several of our endpoints we did not find any issue in the network or on the SIP end.
I've sent some logs to logs@flashphoner.com address.
1. This is the call with problems, found in flashphoner_manager.log.2017-09-26-13 at 13:55:
13:55:33,577 INFO agerRemoteRmiService - RMI TCP Connection(675)-172.31.25.198 SEND REST OBJECT ==>
URL:http://localhost:9091/EchoApp/OnCallEvent
OBJECT:
{
"nodeId" : "XQYpQm0VT8TtFA077nCnME2gYn7xDfKj@172.31.25.198 ",
"appKey" : "defaultApp",
"sessionId" : "/39.x.x.x:61315/172.31.25.198:8443",
"callId" : "421e9d1f0d15200050c17ca81984e5bf@172.31.25.199",
"incoming" : true,
"status" : "PENDING",
"caller" : "xxx0297",
"callee" : "192995",
"hasAudio" : true,
"hasVideo" : false,
"sdp" : "v=0\r\no=root 1090804450 1090804450 IN IP4 172.31.25.199\r\ns=Asterisk PBX 1.6.0.26\r\nc=IN IP4 172.31.25.199\r\nt=0 0\r\nm=audio 10956 RTP/AVP 0 8\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=ptime:20\r\na=sendrecv\r\n",
"visibleName" : "xxx0297_24435_12484_7668_1_1506401724.9370701_",
"mediaProvider" : "WebRTC",
"isMsrp" : false,
"holdForTransfer" : false
}
2. Also there is a .pcap with this exact call. (its the stream between Flashphoner server and Asterisk PBX). The RTP stream seems to have Jitter..but I don't know if this Jitter is due to the connection between Client browser and WCS.
3. Version we are using is : FlashphonerWebCallServer-5.0.1994
Any help is highly appreciated.
