I’m doing tests increasing the ptime for some mobile endpoints. When opus isn’t involved, the values on the SDP seem to be set correctly (so far, tested with g722, alaw, ulaw and gsm). However, once the codec opus is enabled (on the build options and then set to the PJSIP endpoint), I’m observing that maxptime is being always set to 20, which is smaller than ptime I was testing (examples: 40, 60, 120).
This is happening with a default codec opus and custom settings added on codecs.conf.
Firefox with JSSIP seems to deal with this issue by using the ptime sent by Asterisk and, seemingly, ignoring the smaller maxptime. Zoiper on Android, however, has some issues. With Zoiper receiving less packets and sending more, the audio was still fine on both sides, but Asterisk was flooded with the following error. The smaller the value (but still above 20), the more often the error happened:
ERROR[2431029][C-00000004]: codec_opus.c:386 opus_dec_decode: Opus: decoding: buffer too small
Here is an example INVITE, from Asterisk to the JSSIP client (IPs and ports were replaced), showing the maxptime smaller than ptime.
INVITE sip:caracteresAleatorios@ip_LOCAL:portaFora;transport=WS SIP/2.0
Via: SIP/2.0/WS ip_LOCAL:portaServidor;rport;branch=maisUUID;alias
From: “destino” <sip:destino@oNomeDoServidor>;tag=tudoUUID
To: <sip:caracteresAleatorios@ip_LOCAL>
Contact: <sip:asterisk@oNomeDoServidor:portaSIP;transport=ws>
Call-ID: UUID-da-chamada
CSeq: 20023 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 1053v=0
o=- 1301893421 1301893421 IN IP4 ip_INTERNET
s=Asterisk
c=IN IP4 ip_INTERNET
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio portaAudioAqui RTP/SAVPF 107 3 9 8 0 101 102
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 textoLongoUmMonteDePoisPontos
a=ice-ufrag:umTexto
a=ice-pwd:maisTexto
a=candidate:Hbffcb20e 1 UDP 2130706431 ip_INTERNET portaAudioAqui typ host
a=candidate:Hac100011 1 UDP 2130706431 ip_LOCAL portaAudioAqui typ host
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:40
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:numeroAqui cname:umUUID
a=msid:outroUUID maisUmUUID
The template that sets the audio settings for both Zoiper and JSSIP is (the JSSIP endpoint also sets webrtc=yes):
[base](!)
type=endpoint
allow=!all,opus:40,g722:40,alaw:40,ulaw:40,gsm:40
trust_id_outbound=yes
device_state_busy_at=1
dtmf_mode=rfc4733
rtp_symmetric=yes
direct_media=no
force_rport=yes
rewrite_contact=yes
rtp_keepalive=10
rtp_timeout=360
rtp_timeout_hold=36000
rtcp_mux=yes
media_encryption=sdes
force_avp=yes
media_use_received_transport=yes
message_context=recebe-mensagem
set_var=contato_para_chat=1
use_ptime=yes
This was tested on Asterisk 22.3.