[Freeswitch-users] Codecs (Again!)

Andries Venter apventer at gmail.com
Thu Apr 2 11:05:33 MSD 2015


I tried the settings for late negotiation without success (before I mailed
the list!).  I had an action application="answer" statement in my
dialplan.  This seems to override the whole negotiation process!  After I
removed that, the negotiation is done properly.
Thank you Ahmed for the suggestion, even if it did not really solve the
problem.

On Wed, Apr 1, 2015 at 11:48 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> 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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>>
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>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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>
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