Tom Browning
2009-Oct-23 20:26 UTC
[asterisk-users] SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all ; First disallow all codecs allow=siren14 ; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP/AVP 109 34 96 31. b=TIAS:384000. a=rtpmap:109 H264/90000. a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792; max-br=1025. a=rtpmap:34 H263/90000. a=fmtp:34 CIF4=1;CIF=1; Thanks in advance for any tips, Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091023/da25a8b1/attachment.htm
Polycom has a softphone? Is it any good? I've never seen it on their site before. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Browning Sent: Friday, October 23, 2009 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIREN14 call setup and record/playback I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all ; First disallow all codecs allow=siren14 ; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP/AVP 109 34 96 31. b=TIAS:384000. a=rtpmap:109 H264/90000. a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792; max-br=1025. a=rtpmap:34 H263/90000. a=fmtp:34 CIF4=1;CIF=1; Thanks in advance for any tips, Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091023/df66222e/attachment.htm
Kevin P. Fleming
2009-Oct-23 21:01 UTC
[asterisk-users] SIREN14 call setup and record/playback
Tom Browning wrote:> > I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of > Asterisk and I'm trying to get it to accept a SIREN14 call from > Polycom's softphone. Having trouble with SDP negotiation, I want to > only allow SIREN14 and nothing else. I also want to record and playback > files, any tips on what the Record function parameters should be?First, don't enable any codecs labeled 'SIREN7' or 'SIREN14' on the Polycom phone; those are pre-standard names, and they work slightly differently than the ITU standardized codecs.> In sip.conf I have: > > disallow=all ; First disallow all codecs > allow=siren14 ; Is this the right name?Yes, this is correct. Asterisk would also accept 'g.7221c', which is the ITU standardized name.> And the INVITE comes from the Polycom softphone with an SDP of: > > ... > User-Agent: Polycom VV 8.0.4.4035. > ... > m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. > a=rtpmap:99 SIREN14/16000. > a=fmtp:99 bitrate=48000. > a=rtpmap:98 SIREN14/16000. > a=fmtp:98 bitrate=32000. > a=rtpmap:97 SIREN14/16000. > a=fmtp:97 bitrate=24000.These three are all actually Siren7 (16kHz sample rate), but the phone is offering them three times... I've just emailed Polycom about another one of their phones doing this as well. These should all go away if you disable 'Siren14' in the Polycom phone configuration.> a=rtpmap:102 G7221/16000. > a=fmtp:102 bitrate=32000. > a=rtpmap:101 G7221/16000. > a=fmtp:101 bitrate=24000. > a=rtpmap:103 G7221/16000. > a=fmtp:103 bitrate=16000.These are also Siren7 (G.722.1, 16kHz sample rate), at various bit rates. Asterisk supports the 32kbps bit rate, so if you had 'allow=g.7221' or 'allow=siren7' in sip.conf, the first of these options would be accepted. What you want to see in the SDP for ITU G.722.1C is 'G7221/32000' with a bitrate of 48000. If you can find a phone configuration that results that SDP offer, you'll be good to go. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpfleming at digium.com Check us out at www.digium.com & www.asterisk.org
Tom Browning
2009-Nov-04 22:24 UTC
[asterisk-users] SIREN14 call setup and record/playback
Continuing the siren14 usage thread: sip.conf has: disallow=all ; First disallow all codecs allow=siren14 ; Should I be able to originate an outbound call with siren14 as my only codec? When I try originate using either the spool file or a CLI originate command I get: [Nov 4 17:21:49] WARNING[28427]: chan_sip.c:5722 sip_call: No audio format found to offer. Cancelling call to blahblah Inbound calls, record and playback work just great. Now I want to reach out with SIREN14 Thanks in advance, Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091104/952eacde/attachment.htm