Hello, when our Freeswitch sends an INVITE to a phone, it only offers speex codec for voice: m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 When the phone then answers with ACK and OK and sends back it's own codecs during OK, then codecs are negociated to PCMA, which is fine. However our Aastra phones do not like Speex in the INVITE message, so they send back a "Bad request". We have set in vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,GSM,speex@8000h@20i,speex@16000h@20i"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU,G729,GSM,speex@8000h@20i,speex@16000h@20i"/> and the internal profile contains <param name="inbound-codec-prefs" value="PCMA,PCMU,speex,GSM,G729"/> <param name="outbound-codec-prefs" value="PCMA,PCMU,speex,GSM,G729"/> <param name="inbound-codec-negotiation" value="greedy"/> <param name="outbound-codec-negotiation" value="greedy"/> So Speex is not the preferred codec defined. Eeven 'scrooge' did not solve the problem. How can I get rid of this behaviour and force Freeswitch to invite with PCMA? -- With kind regards Peter Steinbach Telefaks Services GmbH Theo-Geisel-Strasse 25 D 61250 Usingen, Germany mailto:lists (att) telefaks.de Internet: www.telefaks.de
You are making the wrong conclusion. The SDP is being verbose. That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged. See the codec list in the m= line? /b On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
Hello,
when our Freeswitch sends an INVITE to a phone, it only offers speex codec for voice: m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
Thanks Brian, that did the trick. Best regards Peter Brian West schrieb:
You are making the wrong conclusion. The SDP is being verbose. That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN
What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged. See the codec list in the m= line?
/b
On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
Hello,
when our Freeswitch sends an INVITE to a phone, it only offers speex codec for voice: m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
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-- With kind regards Peter Steinbach Telefaks Services GmbH Theo-Geisel-Strasse 25 D 61250 Usingen, Germany mailto:lists (att) telefaks.de Internet: www.telefaks.de
Just to be clear in case you're new to SDP... The m=audio line gives a list of codecs identified by by number. Many of them are static numbers meaning they are assigned to particular codecs. The list is at http://www.iana.org/assignments/rtp-parameters 0=PCMU 8=PCMA etc. Anything in the 96-127 range is dynamic and identified by name on a a=rtpmap line. For static numbers it's allowed to have a a=rtpmap line, but not required. But some devices are broken and require it anyway. verbose=sdp=true puts a=rtpmap for all codecs, which will work on all devices but at the risk of possibly exceeding the MTU. -Steve On 11 January 2011 22:54, Brian West <brian@freeswitch.org> wrote:
You are making the wrong conclusion. The SDP is being verbose. That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN
What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged. See the codec list in the m= line?
/b
On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
Hello,
when our Freeswitch sends an INVITE to a phone, it only offers speex codec for voice: m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
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participants (3)
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Brian West
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Peter Steinbach
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Steven Ayre