Hi, Team
How to make RTPEngine transcode the sampling time from 10 milliseconds to 20
milliseconds?
OS: Debian 12
Kamailio ver: 6.0.4
RTPEngine ver: mr13.5.2.0
UAC (ptime:40) --- Kamalio/RTPEngine ---- UAS（ptime:20）
kamailio.cfg:
route[NATMANAGE] {
...
if(has_totag()) {
rtpengine_manage("SIP-source-address replace-origin
replace-session-connection");
} else {
rtpengine_manage("SIP-source-address replace-origin
replace-session-connection codec-mask-all codec-transcode-PCMA;ptime--20
always-transcode");
}
}
rtpengine.log:
```
[1764059397.148615] INFO: [[email protected]]:
[control] Received command 'offer' from 127.0.0.1:49176
[1764059397.148641] DEBUG: [[email protected]]:
[control] Dump for 'offer' from 127.0.0.1:49176: { "supports": [ "load limit"
], "sdp": "v=0
o=1001 1764055864 1764059397 IN IP4 192.168.118.35
s=Kapanga [1764055864]
c=IN IP4 192.168.118.35
t=0 0
m=audio 5112 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=sendrecv
a=maxptime:40
a=ptime:40
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,36
", "flags": [ "SIP-source-address", "always-transcode" ], "replace": [
"origin", "session-connection" ], "codec": { "mask": [ "all" ], "transcode": [
"PCMA;ptime--20" ] }, "call-id":
"[email protected]", "received-from": [ "IP4",
"192.168.118.35" ], "from-tag": "9A4E71D6E3F7B60DB47A7F494D93454F", "command":
"offer" }
[1764059397.148677] INFO: [[email protected]]:
[core] replace-session-connection flag encountered, but not supported anymore.
[1764059397.148694] NOTICE: [[email protected]]:
[core] Creating new call
[1764059397.148801] DEBUG: [[email protected]]:
[core] No matching media (index: 1) using subscription, just use an index.
[1764059397.148824] DEBUG: [[email protected]]:
[core] Subscribing media with monologue tag '' (index: 1) to media with
monologue tag '9A4E71D6E3F7B60DB47A7F494D93454F' (index: 1)
[1764059397.148838] DEBUG: [[email protected]]:
[core] Subscribing media with monologue tag '9A4E71D6E3F7B60DB47A7F494D93454F'
(index: 1) to media with monologue tag '' (index: 1)
[1764059397.148850] DEBUG: [[email protected]]:
[core] setting other slot to 0, setting slot to 0
[1764059397.148857] DEBUG: [[email protected]]:
[codec] Updating codecs for offerer 9A4E71D6E3F7B60DB47A7F494D93454F #1
[1764059397.148877] DEBUG: [[email protected]]:
[codec] Adding codec pcma/8000/ (8)
[1764059397.148886] DEBUG: [[email protected]]:
[codec] Adding codec telephone-event/8000/0-15,36 (101)
[1764059397.148894] DEBUG: [[email protected]]:
[codec] Accepting codec pcma/8000/ (8)
[1764059397.148900] DEBUG: [[email protected]]:
[codec] Accepting codec telephone-event/8000/0-15,36 (101)
[1764059397.148906] DEBUG: [[email protected]]:
[codec] Updating offer codecs for answerer #1
[1764059397.148912] DEBUG: [[email protected]]:
[codec] Adding codec pcma/8000/ (8)
[1764059397.148919] DEBUG: [[email protected]]:
[codec] Adding codec telephone-event/8000/0-15,36 (101)
[1764059397.148929] DEBUG: [[email protected]]:
[codec] Stripping codec pcma/8000/ (8) due to strip=all or strip=full
[1764059397.148944] DEBUG: [[email protected]]:
[codec] Stripping codec telephone-event/8000/0-15,36 (101) due to strip=all or
strip=full
[1764059397.148954] DEBUG: [[email protected]]:
[codec] Adding codec PCMA;ptime=20 for transcoding
[1764059397.148962] WARNING: [[email protected]]:
[core] Usage error: List of codecs empty. Restoring original list of codecs.
Results may be unexpected.
[1764059397.148985] DEBUG: [[email protected]]:
[codec] Adding codec pcma/8000/ (8)
[1764059397.149001] DEBUG: [[email protected]]:
[codec] Adding codec telephone-event/8000/0-15,36 (101)
[1764059397.149005] DEBUG: [[email protected]]:
[codec] Updating supplemental codecs for #1
[1764059397.149017] DEBUG: [[email protected]]:
[codec] Setting up codec handlers for #1 -> 9A4E71D6E3F7B60DB47A7F494D93454F #1
[1764059397.149023] DEBUG: [[email protected]]:
[codec] Default sink codec is pcma/8000/ (8)
[1764059397.149026] DEBUG: [[email protected]]:
[codec] Checking receiver codec pcma/8000/1/ (8)
[1764059397.149028] DEBUG: [[email protected]]:
[codec] Creating codec handler for pcma/8000/ (8)
[1764059397.149036] DEBUG: [[email protected]]:
[codec] Sink codec for pcma/8000/ is pcma/8000/1/ (8)
[1764059397.149040] DEBUG: [[email protected]]:
[codec] Sink supports codec pcma/8000/ (8) for passthrough (to 8)
[1764059397.149046] DEBUG: [[email protected]]:
[codec] Shutting down codec handler for pcma/8000/1
[1764059397.149052] DEBUG: [[email protected]]:
[codec] Using passthrough handler for pcma/8000/ (8) with DTMF 101, CN -1
[1764059397.149067] DEBUG: [[email protected]]:
[codec] Checking receiver codec telephone-event/8000/1/0-15,36 (101)
[1764059397.149073] DEBUG: [[email protected]]:
[codec] Creating codec handler for telephone-event/8000/0-15,36 (101)
[1764059397.149078] DEBUG: [[email protected]]:
[codec] Sink codec for telephone-event/8000/0-15,36 is
telephone-event/8000/1/0-15,36 (101)
[1764059397.149090] DEBUG: [[email protected]]:
[codec] Sink supports codec telephone-event/8000/0-15,36 (101) for passthrough
(to 101)
[1764059397.149093] DEBUG: [[email protected]]:
[codec] Shutting down codec handler for telephone-event/8000/1
[1764059397.149103] DEBUG: [[email protected]]:
[codec] Using passthrough handler for telephone-event/8000/0-15,36 (101) with
DTMF 101, CN -1
[1764059397.149115] DEBUG: [[email protected]]:
[codec] Updating supplemental codecs for #1
[1764059397.149130] DEBUG: [[email protected]]:
[codec] Setting up codec handlers for #1 -> 9A4E71D6E3F7B60DB47A7F494D93454F #1
[1764059397.149145] DEBUG: [[email protected]]:
[codec] Default sink codec is pcma/8000/ (8)
[1764059397.149156] DEBUG: [[email protected]]:
[codec] Checking receiver codec pcma/8000/1/ (8)
[1764059397.149167] DEBUG: [[email protected]]:
[codec] Sink codec for pcma/8000/ is pcma/8000/1/ (8)
[1764059397.149176] DEBUG: [[email protected]]:
[codec] Sink supports codec pcma/8000/ (8) for passthrough (to 8)
[1764059397.149190] DEBUG: [[email protected]]:
[codec] Shutting down codec handler for pcma/8000/1
[1764059397.149207] DEBUG: [[email protected]]:
[codec] Using passthrough handler for pcma/8000/ (8) with DTMF 101, CN -1
[1764059397.149228] DEBUG: [[email protected]]:
[codec] Checking receiver codec telephone-event/8000/1/0-15,36 (101)
[1764059397.149236] DEBUG: [[email protected]]:
[codec] Sink codec for telephone-event/8000/0-15,36 is
telephone-event/8000/1/0-15,36 (101)
[1764059397.149246] DEBUG: [[email protected]]:
[codec] Sink supports codec telephone-event/8000/0-15,36 (101) for passthrough
(to 101)
[1764059397.149252] DEBUG: [[email protected]]:
[codec] Shutting down codec handler for telephone-event/8000/1
[1764059397.149256] DEBUG: [[email protected]]:
[codec] Using passthrough handler for telephone-event/8000/0-15,36 (101) with
DTMF 101, CN -1
[1764059397.149286] DEBUG: [[email protected]]:
[core] Trying to find RTP/RTCP ports (allocation attempt = '0')
[1764059397.149302] DEBUG: [[email protected]]:
[core] Trying to bind the socket for RTP/RTCP ports (allocation attempt = '1')
[1764059397.149306] DEBUG: [[email protected]]:
[core] Trying to bind the socket for port = '10068'
[1764059397.149439] DEBUG: [[email protected]]:
[core] Trying to bind the socket for port = '10069'
[1764059397.149464] DEBUG: [[email protected]]:
[core] Opened 2 socket(s) from port '10068' (on interface '192.168.118.57') for
a media relay
[1764059397.149533] DEBUG: [[email protected]]:
[core] Trying to find RTP/RTCP ports (allocation attempt = '0')
[1764059397.149544] DEBUG: [[email protected]]:
[core] Trying to bind the socket for RTP/RTCP ports (allocation attempt = '1')
[1764059397.149547] DEBUG: [[email protected]]:
[core] Trying to bind the socket for port = '10000'
[1764059397.149561] DEBUG: [[email protected]]:
[core] Trying to bind the socket for port = '10001'
[1764059397.149570] DEBUG: [[email protected]]:
[core] Opened 2 socket(s) from port '10000' (on interface '192.168.118.57') for
a media relay
[1764059397.149594] DEBUG: [[email protected]]:
[core] set FILLED flag for stream, local 192.168.118.57:10000 remote
192.168.118.35:5112
[1764059397.149602] DEBUG: [[email protected]]:
[core] set FILLED flag for stream, local 192.168.118.57:10001 remote
192.168.118.35:5113
[1764059397.149637] INFO: [[email protected]]:
[control] Replying to 'offer' from 127.0.0.1:49176 (elapsed time 0.000980 sec)
[1764059397.149653] DEBUG: [[email protected]]:
[control] Response dump for 'offer' to 127.0.0.1:49176: { "sdp": "v=0
o=1001 1764055864 1764059397 IN IP4 192.168.118.57
s=Kapanga [1764055864]
t=0 0
m=audio 10068 RTP/AVP 8 101
c=IN IP4 192.168.118.57
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,36
a=sendrecv
a=rtcp:10069
a=ptime:40
a=maxptime:40
", "result": "ok" }
```
codec-transcode-PCMA;ptime--20 doesn't seem to work. The response for 'offer'
contains `ptime:40`.
What I really want is to be able to pass 'ptime=20' in the flags, but the
rtpengine module doesn't seem to support it.
Thanks!
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