[Freeswitch-users] Codecs (Again!)
Ahmed Sboor
ahmed at netelsat.net
Thu Apr 2 01:48:04 MSD 2015
Hi Andries ,
You should again look on https://wiki.freeswitch.org/wiki/Codec_Negotiation
And i think easiest way is using *absolute_codec_string *before briding to
"G729" only PABX.
Ahmed
On Wed, Apr 1, 2015 at 7:51 PM, Andries Venter <apventer at gmail.com> wrote:
> I am using Freeswitch (1.4.8) as an SBC between our company PABX and our
> ISP's Trunk (also terminating on an SBC).
>
> My problem is that I need G729 in some cases and PCMA in others (the
> majority).
>
>
>
> The setup looks more or less like this:
>
> Our PABX (192.168.102.215)=>(192.168.102.102)Our SBC
> (10.17.159.10)=>(10.17.159.9)ISP SBC=>Rest of the world
>
>
>
> Since we use private addresses, Freeswitch must do RTP proxying. No
> transcoding must be done!!!
>
>
>
> I have set global_codec_prefs to "PCMA,G729". Then everything to
> destination supporting PCMA works. When trying to connect to a G729-only
> PABX, the call fails. Leg A is setup as follows
>
>
>
> 192.168.102.215.5060 > 192.168.102.102.5090: SIP, length: 933
>
> INVITE sip:0873611337 at 192.168.102.102:5090;transport=udp SIP/2.0
>
> To: <sip:0873611337 at 192.168.102.102;user=phone>
>
> From: "JGD Winson" <sip:anonymous at anonymous.invalid
> >;tag=snl_iFGEOrlLLf
>
> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo
>
> CSeq: 1235 INVITE
>
> Contact: <sip:anonymous at 192.168.102.215:5060;maddr=192.168.102.215
> >
>
> Via: SIP/2.0/UDP 192.168.102.215:5060
> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk
>
> Content-Type: application/sdp
>
> Content-Length: 288
>
> X-Siemens-Call-Type: ST-insecure
>
> Accept-Language: en;q=0.0
>
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
>
> Date: Wed, 01 Apr 2015 14:30:15 GMT
>
> Max-Forwards: 69
>
>
>
> v=0
>
> o=MxSIP 0 105140472 IN IP4 10.11.32.226
>
> s=SIP Call
>
> c=IN IP4 10.11.32.226
>
> t=0 0
>
> m=audio 5014 RTP/AVP 8 0 18 101
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=silenceSupp:off - - - -
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-15
>
>
>
> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 396
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 192.168.102.215:5060
> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk
>
> From: "JGD Winson" <sip:anonymous at anonymous.invalid
> >;tag=snl_iFGEOrlLLf
>
> To: <sip:0873611337 at 192.168.102.102;user=phone>
>
> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo
>
> CSeq: 1235 INVITE
>
> User-Agent:
> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>
> Content-Length: 0
>
>
>
> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 1201
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 192.168.102.215:5060
> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk
>
> From: "JGD Winson" <sip:anonymous at anonymous.invalid
> >;tag=snl_iFGEOrlLLf
>
> To: <sip:0873611337 at 192.168.102.102;user=phone>;tag=BytX6X1t7HytF
>
> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo
>
> CSeq: 1235 INVITE
>
> Contact: <sip:0873611337 at 192.168.102.102:5090;transport=udp>
>
> User-Agent:
> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>
> Accept: application/sdp
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, path, replaces
>
> Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>
> Content-Type: application/sdp
>
> Content-Disposition: session
>
> Content-Length: 226
>
> Remote-Party-ID: "0873611337" <sip:0873611337 at 192.168.102.102
> >;party=calling;privacy=off;screen=no
>
>
>
> v=0
>
> o=FreeSWITCH 1427881297 1427881298 IN IP4 192.168.102.102
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.102.102
>
> t=0 0
>
> m=audio 17318 RTP/AVP 8 101
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
>
>
>
>
> When I change my global_codec_prefs to "G729,PCMA", G729 is offered to our
> PABX by Freeswitch, then to the ISP and then it works to destinations
> supporting G729. It fails to destinations not supporting G729. Now Leg
> A is setup as follows:
>
>
>
> 192.168.102.215.5060 > 192.168.102.102.5090: SIP, length: 934
>
> INVITE sip:0873611337 at 192.168.102.102:5090;transport=udp SIP/2.0
>
> To: <sip:0873611337 at 192.168.102.102;user=phone>
>
> From: "JGD Winson" <sip:anonymous at anonymous.invalid
> >;tag=snl_8NZMvB3KNA
>
> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm
>
> CSeq: 1235 INVITE
>
> Contact: <sip:anonymous at 192.168.102.215:5060;maddr=192.168.102.215
> >
>
> Via: SIP/2.0/UDP 192.168.102.215:5060
> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0
>
> Content-Type: application/sdp
>
> Content-Length: 289
>
> X-Siemens-Call-Type: ST-insecure
>
> Accept-Language: en;q=0.0
>
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
>
> Date: Wed, 01 Apr 2015 14:27:51 GMT
>
> Max-Forwards: 69
>
>
>
> v=0
>
> o=MxSIP 0 1224193414 IN IP4 10.11.32.226
>
> s=SIP Call
>
> c=IN IP4 10.11.32.226
>
> t=0 0
>
> m=audio 5014 RTP/AVP 8 0 18 101
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=silenceSupp:off - - - -
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-15
>
>
>
> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 396
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 192.168.102.215:5060
> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0
>
> From: "JGD Winson" <sip:anonymous at anonymous.invalid
> >;tag=snl_8NZMvB3KNA
>
> To: <sip:0873611337 at 192.168.102.102;user=phone>
>
> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm
>
> CSeq: 1235 INVITE
>
> User-Agent:
> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>
> Content-Length: 0
>
>
>
> 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 1224
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 192.168.102.215:5060
> ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0
>
> From: "JGD Winson" <sip:anonymous at anonymous.invalid
> >;tag=snl_8NZMvB3KNA
>
> To: <sip:0873611337 at 192.168.102.102;user=phone>;tag=g2eKXDUB6y4Hr
>
> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm
>
> CSeq: 1235 INVITE
>
> Contact: <sip:0873611337 at 192.168.102.102:5090;transport=udp>
>
> User-Agent:
> FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
>
> Accept: application/sdp
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Supported: timer, path, replaces
>
> Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>
> Content-Type: application/sdp
>
> Content-Disposition: session
>
> Content-Length: 249
>
> Remote-Party-ID: "0873611337" <sip:0873611337 at 192.168.102.102
> >;party=calling;privacy=off;screen=no
>
>
>
> v=0
>
> o=FreeSWITCH 1427866745 1427866746 IN IP4 192.168.102.102
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.102.102
>
> t=0 0
>
> m=audio 31726 RTP/AVP 18 101
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
>
>
> I have tried various (hopefully all) combinations of
> inbound-codec-negotiation (generous, greedy and scrooge) and
> inbound-late-negotiation (true, false). Nothing does what I want.
>
>
>
> I believe Freeswitch needs to negotiate the codecs with the endpoint and
> choose one from my list. However, it seems to grab the first one in my
> list before checking the endpoint. Then it offers only that codec to the
> endpoint, which of course ignores / rejects it!
>
>
>
> What can I do?
>
> Regards
> Andries
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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