INCOMPATIBLE_DESTINATION - did FS not offer a codec?
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs? FS log: http://pastebin.freeswitch.org/17316 PCAP: http://ge.tt/95tFji7 Can someone tell me what's going on here? -Avi
Try setting verbose SDP: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp On Tue, Sep 13, 2011 at 11:01 AM, Avi Marcus <avi@avimarcus.net> wrote:
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs? FS log: http://pastebin.freeswitch.org/17316 PCAP: http://ge.tt/95tFji7 Can someone tell me what's going on here? -Avi
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-- Kristian Kielhofner
For the record, line #107 contains the codecs (and other stuff) that FS offered: m=audio 16480 RTP/AVP 8 0 9 101 13 The "8 0 9 101 13" represent the codecs (and other stuff). What do those numbers mean? This wiki page has the list: http://wiki.freeswitch.org/wiki/RTP_payload_list The SDP spec says that for non-dynamic payload numbers you are NOT required to have the "a=rtpmap" lines. This makes sense because those payload numbers don't change meaning until you get up to 96 and higher. The issue is that some devices are broken and require the "a=rtpmap" lines for the payload types below number 96. The verbose_sdp chan var mentioned by KK tells FS to add all those a=rtpmap lines for the non-dynamic payload types. You should only set this var when absolutely necessary because those verbose SDP's can cause the packets to be larger than the allowable MTU size. Dontcha just *love* SIP/SDP interop? :) -MC On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi@avimarcus.net> wrote:
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs? FS log: http://pastebin.freeswitch.org/17316 PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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OK then! Looking at my call logs, it seems to mostly happen with one particular carrier. Can I set verbose_sdp=true in the bridge string? e.g. "[verbose_sdp=true]sofia/gateway/... "? Thanks, -Avi On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc@freeswitch.org> wrote:
For the record, line #107 contains the codecs (and other stuff) that FS offered:
m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those numbers mean? This wiki page has the list:
http://wiki.freeswitch.org/wiki/RTP_payload_list
The SDP spec says that for non-dynamic payload numbers you are NOT required to have the "a=rtpmap" lines. This makes sense because those payload numbers don't change meaning until you get up to 96 and higher. The issue is that some devices are broken and require the "a=rtpmap" lines for the payload types below number 96. The verbose_sdp chan var mentioned by KK tells FS to add all those a=rtpmap lines for the non-dynamic payload types. You should only set this var when absolutely necessary because those verbose SDP's can cause the packets to be larger than the allowable MTU size.
Dontcha just *love* SIP/SDP interop? :)
-MC
On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi@avimarcus.net> wrote:
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs? FS log: http://pastebin.freeswitch.org/17316 PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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I believe that is exactly the way to use it. -MC On Tue, Sep 13, 2011 at 4:53 PM, Avi Marcus <avi@avimarcus.net> wrote:
OK then! Looking at my call logs, it seems to mostly happen with one particular carrier. Can I set verbose_sdp=true in the bridge string? e.g. "[verbose_sdp=true]sofia/gateway/... "?
Thanks, -Avi
On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc@freeswitch.org>wrote:
For the record, line #107 contains the codecs (and other stuff) that FS offered:
m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those numbers mean? This wiki page has the list:
http://wiki.freeswitch.org/wiki/RTP_payload_list
The SDP spec says that for non-dynamic payload numbers you are NOT required to have the "a=rtpmap" lines. This makes sense because those payload numbers don't change meaning until you get up to 96 and higher. The issue is that some devices are broken and require the "a=rtpmap" lines for the payload types below number 96. The verbose_sdp chan var mentioned by KK tells FS to add all those a=rtpmap lines for the non-dynamic payload types. You should only set this var when absolutely necessary because those verbose SDP's can cause the packets to be larger than the allowable MTU size.
Dontcha just *love* SIP/SDP interop? :)
-MC
On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi@avimarcus.net> wrote:
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs? FS log: http://pastebin.freeswitch.org/17316 PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You should also be able to set it in the gateway configuration, I believe: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables If that works then that'll probably make it far easier to manage as you won't need to put in any logic to alter the dialstring. -Steve On 14 September 2011 00:53, Avi Marcus <avi@avimarcus.net> wrote:
OK then! Looking at my call logs, it seems to mostly happen with one particular carrier. Can I set verbose_sdp=true in the bridge string? e.g. "[verbose_sdp=true]sofia/gateway/... "?
Thanks, -Avi
On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc@freeswitch.org>wrote:
For the record, line #107 contains the codecs (and other stuff) that FS offered:
m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those numbers mean? This wiki page has the list:
http://wiki.freeswitch.org/wiki/RTP_payload_list
The SDP spec says that for non-dynamic payload numbers you are NOT required to have the "a=rtpmap" lines. This makes sense because those payload numbers don't change meaning until you get up to 96 and higher. The issue is that some devices are broken and require the "a=rtpmap" lines for the payload types below number 96. The verbose_sdp chan var mentioned by KK tells FS to add all those a=rtpmap lines for the non-dynamic payload types. You should only set this var when absolutely necessary because those verbose SDP's can cause the packets to be larger than the allowable MTU size.
Dontcha just *love* SIP/SDP interop? :)
-MC
On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi@avimarcus.net> wrote:
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs? FS log: http://pastebin.freeswitch.org/17316 PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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participants (4)
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Avi Marcus
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Kristian Kielhofner
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Michael Collins
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Steven Ayre