Codec mismatch IP Phone

[murraymwps]

We have a hosted phone service and have been seeing audio dropouts occasionally. With a packet sniffer I confirmed that packets weren't being dropped in either direction but I did see a codec mismatch. On one "problem" call (the only one I've captured so far) the IP phone was sending audio using G711. The far end was sending using G729. The far end could hear the IP phone user but the IP phone user couldn't hear the far end. My guess is that the G711 stream is being converted to G729 somehow so the far end can understand it but the IP phone expects to receive G711. Does that sound right or should an IP phone be able to send G711 and receive G729?

Thanks,

-mike

[Rajan]

Hi Mike,

IP Phones can support multiple codec but for one call it can speak only one codec which is being negotiated. In case the other end wants to use a different codec, transcoders will be invoked for codec conversion.

HTH

Shenbagarajan

[Vivek Batra]

This is unlikely in the real world cases afaik and I haven't see any implementation across different vendors supporting sending and receiving RTP using different codec's. 

Although if you ask it's technically possible, I would say 'Yes'. If we look at SIP, it's possible to negotiate more than one codec with in a SIP dialog and in such case, it's possible for UA to switch between the codec's without renegotiating in Re-INVITE. Now why UA needs to switch from one codec to other, let say from G711 to G729 could be b/c of UA can detect packet loss during call by means of RTCP statistics. 

[Brandon Pierce]

Transcoders are responsible for converting from one codec to another.  What do the SIP logs show?  You might have better luck looking at the messages assuming you are using SIP.