Benoît Panizzon
2022-May-20 14:37 UTC
[asterisk-users] Asterisk chan_sip removing ptime from DSP?
Hi List Just ran into another weird issue... In Swiss Telephone Interconnection, ptime=20 is a requirement. So on our SBC we enforce the presence of ptime=20 to avoid issues. I have an asterisk with chan_sip in the LAB which behaves weirdly... Inbound SDP audio part: m=audio 15542 RTP/AVP 9 8 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:4hJMPvZmRA03KxLg8Hp+aHqeIhnmYBSQtwlT+Vkr a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Outbound SDP audio part m=audio 18802 RTP/AVP 8 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv Why is ptime missing on the outbound leg? Our SBC answers 415 -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________
Benoît Panizzon
2022-May-20 15:00 UTC
[asterisk-users] Asterisk chan_sip removing ptime from DSP?
Also figured that out. For chan sip, specifying the ptime in the allow statement allow=g722:20,alaw:20 made the ptime header be present and the SBC happy. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________