Actually we had same issues with few soft switches and we resolved it by using separate  profiles at all , though absolute codec string also worked perfect . Am Glad its done and thanks for sharing the cause of problem.


On Thu, Apr 2, 2015 at 12:05 PM, Andries Venter <apventer@gmail.com> wrote:
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@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@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@192.168.102.102:5090;transport=udp SIP/2.0

        To: <sip:0873611337@192.168.102.102;user=phone>

        From: "JGD Winson" <sip:anonymous@anonymous.invalid>;tag=snl_iFGEOrlLLf

        Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo

        CSeq: 1235 INVITE

        Contact: <sip:anonymous@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@anonymous.invalid>;tag=snl_iFGEOrlLLf

        To: <sip:0873611337@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@anonymous.invalid>;tag=snl_iFGEOrlLLf

        To: <sip:0873611337@192.168.102.102;user=phone>;tag=BytX6X1t7HytF

        Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo

        CSeq: 1235 INVITE

        Contact: <sip:0873611337@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@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@192.168.102.102:5090;transport=udp SIP/2.0

        To: <sip:0873611337@192.168.102.102;user=phone>

        From: "JGD Winson" <sip:anonymous@anonymous.invalid>;tag=snl_8NZMvB3KNA

        Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm

        CSeq: 1235 INVITE

        Contact: <sip:anonymous@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@anonymous.invalid>;tag=snl_8NZMvB3KNA

        To: <sip:0873611337@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@anonymous.invalid>;tag=snl_8NZMvB3KNA

        To: <sip:0873611337@192.168.102.102;user=phone>;tag=g2eKXDUB6y4Hr

        Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm

        CSeq: 1235 INVITE

        Contact: <sip:0873611337@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@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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