I'm having an issue with the linksys spa devices when enabling inbound proxy media mode (inbound-proxy-media=true) and late negotiation (inbound-late-negotiation=true) in the sofia profile. The spa immediately sends a BYE when the call is answered by the called party. For whatever reason, it works fine between two linksys devices directly connected to FS, but when the call goes out to the PSTN via the SIP provider, the spa isn't happy and sends a BYE. After comparing the raw SIP packets on the wire (tcpdump) and between enabling/disabling proxy-media mode and late negotiation, the only difference I notice is the port in the m= line of the SDP header. According to the freeswitch log, the rtp port would be rewritten to 28044 in the sdp header of the SIP packet sent to the spa device. But on the wire, the port is rewritten to 0, which I'm guessing is why the spa isn't happy and sending a BYE. Here's the excerpt from the freeswitch log showing FS rewriting the port to 28044 for the packet going to the spa device. [DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() sofia/internal/<phone_number_removed>@mydomain.net Patched SDP --- v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 24174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 +++ v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 28044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 However, the packet on the wire reveals FS rewriting the port to 0 ... v=0. o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 0 RTP/AVP 96 101. a=rtpmap:96 G729/8000. a=fmtp:96 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. Is this a bug or is there some other problem? Thanks for the help, Gabe
Which version of FreeSWITCH are you using? /b On Oct 20, 2008, at 2:25 AM, Gabriel Kuri wrote:
I'm having an issue with the linksys spa devices when enabling inbound proxy media mode (inbound-proxy-media=true) and late negotiation (inbound-late-negotiation=true) in the sofia profile. The spa immediately sends a BYE when the call is answered by the called party. For whatever reason, it works fine between two linksys devices directly connected to FS, but when the call goes out to the PSTN via the SIP provider, the spa isn't happy and sends a BYE.
After comparing the raw SIP packets on the wire (tcpdump) and between enabling/disabling proxy-media mode and late negotiation, the only difference I notice is the port in the m= line of the SDP header.
According to the freeswitch log, the rtp port would be rewritten to 28044 in the sdp header of the SIP packet sent to the spa device. But on the wire, the port is rewritten to 0, which I'm guessing is why the spa isn't happy and sending a BYE.
Here's the excerpt from the freeswitch log showing FS rewriting the port to 28044 for the packet going to the spa device.
[DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() sofia/internal/<phone_number_removed>@mydomain.net Patched SDP --- v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 24174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
+++ v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 28044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
However, the packet on the wire reveals FS rewriting the port to 0 ...
v=0. o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 0 RTP/AVP 96 101. a=rtpmap:96 G729/8000. a=fmtp:96 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
Is this a bug or is there some other problem?
Thanks for the help, Gabe
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Sorry, I should've included that in the original mesg... The logs I posted were from trunk-r10055. I updated this morning to trunk-r10081, still the same issue. ~Gabe Brian West wrote:
Which version of FreeSWITCH are you using?
/b
On Oct 20, 2008, at 2:25 AM, Gabriel Kuri wrote:
I'm having an issue with the linksys spa devices when enabling inbound proxy media mode (inbound-proxy-media=true) and late negotiation (inbound-late-negotiation=true) in the sofia profile. The spa immediately sends a BYE when the call is answered by the called party. For whatever reason, it works fine between two linksys devices directly connected to FS, but when the call goes out to the PSTN via the SIP provider, the spa isn't happy and sends a BYE.
After comparing the raw SIP packets on the wire (tcpdump) and between enabling/disabling proxy-media mode and late negotiation, the only difference I notice is the port in the m= line of the SDP header.
According to the freeswitch log, the rtp port would be rewritten to 28044 in the sdp header of the SIP packet sent to the spa device. But on the wire, the port is rewritten to 0, which I'm guessing is why the spa isn't happy and sending a BYE.
Here's the excerpt from the freeswitch log showing FS rewriting the port to 28044 for the packet going to the spa device.
[DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() sofia/internal/<phone_number_removed>@mydomain.net Patched SDP --- v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 24174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
+++ v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 28044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
However, the packet on the wire reveals FS rewriting the port to 0 ...
v=0. o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 0 RTP/AVP 96 101. a=rtpmap:96 G729/8000. a=fmtp:96 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
Is this a bug or is there some other problem?
Thanks for the help, Gabe
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I ran into this posting which is similar, although not exactly the same, as the failure mode I'm experiencing. http://bugs.digium.com/view.php?id=11483 following what this other person tried to temporarily fix the issue, I changed the name of the rtpmap on the linksys spa from G729a to G729 and it works - FS no longer transmits an audio port of 0 in the sdp headers when inbound-proxy-media and late-negotiation are enabled. correct sdp header excerpt on a call ... v=0. o=01Nextone 2341985734634606731 5798373005113647141 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 25454 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. So is this an underlying issue with the linksys spa units or FS? Gabe Gabriel Kuri wrote:
I'm having an issue with the linksys spa devices when enabling inbound proxy media mode (inbound-proxy-media=true) and late negotiation (inbound-late-negotiation=true) in the sofia profile. The spa immediately sends a BYE when the call is answered by the called party. For whatever reason, it works fine between two linksys devices directly connected to FS, but when the call goes out to the PSTN via the SIP provider, the spa isn't happy and sends a BYE.
After comparing the raw SIP packets on the wire (tcpdump) and between enabling/disabling proxy-media mode and late negotiation, the only difference I notice is the port in the m= line of the SDP header.
According to the freeswitch log, the rtp port would be rewritten to 28044 in the sdp header of the SIP packet sent to the spa device. But on the wire, the port is rewritten to 0, which I'm guessing is why the spa isn't happy and sending a BYE.
Here's the excerpt from the freeswitch log showing FS rewriting the port to 28044 for the packet going to the spa device.
[DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() sofia/internal/<phone_number_removed>@mydomain.net Patched SDP --- v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 24174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
+++ v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 28044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
However, the packet on the wire reveals FS rewriting the port to 0 ...
v=0. o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 0 RTP/AVP 96 101. a=rtpmap:96 G729/8000. a=fmtp:96 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
Is this a bug or is there some other problem?
Thanks for the help, Gabe
It's an spa issue but we work around it so that shouldn't matter! /b Sent from my iPhone On Oct 20, 2008, at 9:59 PM, Gabriel Kuri <gkuri@ieee.org> wrote:
I ran into this posting which is similar, although not exactly the same, as the failure mode I'm experiencing.
http://bugs.digium.com/view.php?id=11483
following what this other person tried to temporarily fix the issue, I changed the name of the rtpmap on the linksys spa from G729a to G729 and it works - FS no longer transmits an audio port of 0 in the sdp headers when inbound-proxy-media and late-negotiation are enabled.
correct sdp header excerpt on a call ...
v=0. o=01Nextone 2341985734634606731 5798373005113647141 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 25454 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20.
So is this an underlying issue with the linksys spa units or FS?
Gabe
Gabriel Kuri wrote:
I'm having an issue with the linksys spa devices when enabling inbound proxy media mode (inbound-proxy-media=true) and late negotiation (inbound-late-negotiation=true) in the sofia profile. The spa immediately sends a BYE when the call is answered by the called party. For whatever reason, it works fine between two linksys devices directly connected to FS, but when the call goes out to the PSTN via the SIP provider, the spa isn't happy and sends a BYE.
After comparing the raw SIP packets on the wire (tcpdump) and between enabling/disabling proxy-media mode and late negotiation, the only difference I notice is the port in the m= line of the SDP header.
According to the freeswitch log, the rtp port would be rewritten to 28044 in the sdp header of the SIP packet sent to the spa device. But on the wire, the port is rewritten to 0, which I'm guessing is why the spa isn't happy and sending a BYE.
Here's the excerpt from the freeswitch log showing FS rewriting the port to 28044 for the packet going to the spa device.
[DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() sofia/internal/<phone_number_removed>@mydomain.net Patched SDP --- v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 24174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
+++ v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 28044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
However, the packet on the wire reveals FS rewriting the port to 0 ...
v=0. o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 0 RTP/AVP 96 101. a=rtpmap:96 G729/8000. a=fmtp:96 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
Is this a bug or is there some other problem?
Thanks for the help, Gabe
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participants (2)
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Brian West
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Gabriel Kuri